diff --git a/MakeLegacy.sh b/MakeLegacy.sh
index bb6958e4..3c8e9152 100755
--- a/MakeLegacy.sh
+++ b/MakeLegacy.sh
@@ -1,4 +1,17 @@
 #!/bin/sh
 
+
+#LINE=`grep "AC_INIT(" configure.ac`
+#AC_INIT([mpg123], [0.60-devel], [mpg123-devel@lists.sourceforge.net])
+
+
+
 cd src
+
+# Need to extract this automatically from configure.ac
+echo "// Created by MakeLegacy.sh" > config.h
+echo "#define PACKAGE_NAME \"mpg123\"" >> config.h
+echo "#define PACKAGE_VERSION \"0.60-devel\"" >> config.h
+
+
 exec make -f Makefile.legacy $*
diff --git a/configure.ac b/configure.ac
index f1983760..ca0f2a0b 100644
--- a/configure.ac
+++ b/configure.ac
@@ -124,7 +124,7 @@ AC_CHECK_HEADERS([CoreAudio/CoreAudio.h], HAVE_COREAUDIO=yes, HAVE_COREAUDIO=no)
 
 dnl ############## Choose compiler flags and CPU
 
-CFLAGS="$CFLAGS -DHAVE_CONFIG_H -Wall -std=c99"
+CFLAGS="$CFLAGS -Wall -std=c99"
 
 
 # If debugging is enabled then make warnings errors
diff --git a/src/Makefile.legacy b/src/Makefile.legacy
index 2ea43954..ad8dcf79 100644
--- a/src/Makefile.legacy
+++ b/src/Makefile.legacy
@@ -232,7 +232,7 @@ linux-alpha:
 		CFLAGS="$(CFLAGS) $(CPPFLAGS) -DLINUX -DOSS -DHAVE_TERMIOS -Wall -O2 \
 			-fomit-frame-pointer -funroll-all-loops \
 			-finline-functions -ffast-math \
-			-Wall -O6 -DUSE_MMAP" \
+			-Wall -O6 -DHAVE_MMAP" \
 		mpg123-make
 
 linux-alpha-esd:
@@ -242,7 +242,7 @@ linux-alpha-esd:
 		CFLAGS="$(CFLAGS) $(CPPFLAGS) -DLINUX -DOSS -DHAVE_TERMIOS -Wall -O2 \
 			-fomit-frame-pointer -funroll-all-loops \
 			-finline-functions -ffast-math \
-			-Wall -O6 -DUSE_MMAP" \
+			-Wall -O6 -DHAVE_MMAP" \
 		mpg123-make
 
 #linux-ppc:
@@ -286,7 +286,7 @@ linux-ppc-esd:
 linux-sparc:
 	$(MAKE) CC=gcc \
 		OBJECTS='decode.o dct64.o audio_sun.o term.o' \
-		CFLAGS="$(CFLAGS) $(CPPFLAGS) -DREAL_IS_FLOAT -DUSE_MMAP -DSPARCLINUX -Wall -O2 \
+		CFLAGS="$(CFLAGS) $(CPPFLAGS) -DREAL_IS_FLOAT -DHAVE_MMAP -DSPARCLINUX -Wall -O2 \
 			-DHAVE_TERMIOS \
 			-fomit-frame-pointer -funroll-all-loops \
 			-finline-functions -ffast-math" \
@@ -363,7 +363,7 @@ freebsd:
 		CFLAGS="$(CFLAGS) $(CPPFLAGS) -Wall -ansi -pedantic -O4 -mcpu=i486 -fomit-frame-pointer \
 			-funroll-all-loops -ffast-math -DROT_I386 \
 			-DREAD_MMAP \
-			-DI386_ASSEM -DREAL_IS_FLOAT -DUSE_MMAP -DOSS" \
+			-DI386_ASSEM -DREAL_IS_FLOAT -DHAVE_MMAP -DOSS" \
 		mpg123-make
 
 freebsd-esd:
@@ -373,7 +373,7 @@ freebsd-esd:
 		CFLAGS="$(CFLAGS) $(CPPFLAGS) -Wall -ansi -pedantic -O4 -mcpu=i486 -fomit-frame-pointer \
 			-funroll-all-loops -ffast-math -DROT_I386 \
 			-DREAD_MMAP \
-			-DI386_ASSEM -DREAL_IS_FLOAT -DUSE_MMAP -DOSS \
+			-DI386_ASSEM -DREAL_IS_FLOAT -DHAVE_MMAP -DOSS \
 			-I/usr/local/include -L/usr/local/lib" \
 		mpg123-make
 
@@ -384,7 +384,7 @@ freebsd-frontend:
 		CFLAGS="$(CFLAGS) $(CPPFLAGS) -Wall -ansi -pedantic -O4 -mcpu=i486 -fomit-frame-pointer \
 			-funroll-all-loops -ffast-math -DROT_I386 \
 			-DFRONTEND \
-			-DI386_ASSEM -DREAL_IS_FLOAT -DUSE_MMAP -DOSS" \
+			-DI386_ASSEM -DREAL_IS_FLOAT -DHAVE_MMAP -DOSS" \
 		$(FRONTEND)
  
 
@@ -397,14 +397,14 @@ solaris-pure:
 		LDFLAGS="$(LDFLAGS) -lsocket -lnsl" \
 		OBJECTS='decode.o dct64.o audio_sun.o term.o' \
 		CFLAGS="$(CFLAGS) $(CPPFLAGS) -fast -native -xO4 -DSOLARIS -DHAVE_TERMIOS \
-			-DUSE_MMAP" \
+			-DHAVE_MMAP" \
 		mpg123-make
 
 solaris-ccscc:
 	$(MAKE) CC=/usr/ccs/bin/ucbcc LDFLAGS="$(LDFLAGS) -lsocket -lnsl" \
 		OBJECTS='decode.o dct64.o audio_sun.o term.o' \
 		CFLAGS="$(CFLAGS) $(CPPFLAGS) -fast -native -xO4 -DSOLARIS \
-			-DUSE_MMAP" \
+			-DHAVE_MMAP" \
 		mpg123-make
 
 # common solaris compile entries
@@ -412,14 +412,14 @@ solaris:
 	$(MAKE) CC=cc LDFLAGS="$(LDFLAGS) -lsocket -lnsl" \
 		OBJECTS='decode.o dct64.o audio_sun.o term.o' \
 		CFLAGS="$(CFLAGS)-fast -native -xO4 -DSOLARIS \
-			-DUSE_MMAP -DHAVE_TERMIOS" \
+			-DHAVE_MMAP -DHAVE_TERMIOS" \
 		mpg123-make
 
 solaris-gcc-profile:
 	$(MAKE) CC='gcc' \
 		LDFLAGS="$(LDFLAGS) -lsocket -lnsl -pg" \
 		OBJECTS='decode.o dct64.o audio_sun.o' \
-		CFLAGS="$(CFLAGS) $(CPPFLAGS) -g -pg -O2 -Wall -DSOLARIS -DREAL_IS_FLOAT -DUSE_MMAP \
+		CFLAGS="$(CFLAGS) $(CPPFLAGS) -g -pg -O2 -Wall -DSOLARIS -DREAL_IS_FLOAT -DHAVE_MMAP \
 			-DREAD_MMAP \
 			-funroll-all-loops -finline-functions" \
 		mpg123-make
@@ -428,7 +428,7 @@ solaris-gcc:
 	$(MAKE) CC=gcc \
 		LDFLAGS="$(LDFLAGS) -lsocket -lnsl" \
 		OBJECTS='decode.o dct64.o audio_sun.o term.o' \
-		CFLAGS="$(CFLAGS) $(CPPFLAGS) -O2 -Wall -pedantic -DSOLARIS -DREAL_IS_FLOAT -DUSE_MMAP \
+		CFLAGS="$(CFLAGS) $(CPPFLAGS) -O2 -Wall -pedantic -DSOLARIS -DREAL_IS_FLOAT -DHAVE_MMAP \
 			-DREAD_MMAP -DHAVE_TERMIOS \
 			-funroll-all-loops  -finline-functions" \
 		mpg123-make
@@ -437,7 +437,7 @@ solaris-gcc-esd:
 	$(MAKE) CC=gcc LDFLAGS="$(LDFLAGS) -lsocket -lnsl" \
 		AUDIO_LIB='-lesd -lresolv' \
 		OBJECTS='decode.o dct64.o audio_esd.o' \
-		CFLAGS="$(CFLAGS) $(CPPFLAGS) -O2 -Wall -DSOLARIS -DREAL_IS_FLOAT -DUSE_MMAP \
+		CFLAGS="$(CFLAGS) $(CPPFLAGS) -O2 -Wall -DSOLARIS -DREAL_IS_FLOAT -DHAVE_MMAP \
 			-DREAD_MMAP \
 			-funroll-all-loops -finline-functions" \
 		mpg123-make
@@ -446,7 +446,7 @@ solaris-x86-gcc-oss:
 	$(MAKE) CC=gcc LDFLAGS="$(LDFLAGS) -lsocket -lnsl" \
 		OBJECTS='decode_i386.o dct64_i386.o $(PENTIUM_DECODE).o \
 			audio_oss.o' \
-		CFLAGS="$(CFLAGS) $(CPPFLAGS) -DI386_ASSEM -DREAL_IS_FLOAT -DPENTIUM_OPT -DUSE_MMAP \
+		CFLAGS="$(CFLAGS) $(CPPFLAGS) -DI386_ASSEM -DREAL_IS_FLOAT -DPENTIUM_OPT -DHAVE_MMAP \
 			-DREAD_MMAP -DOSS \
 			-Wall -O2 -mcpu=i486 \
 			-funroll-all-loops -finline-functions" \
@@ -457,7 +457,7 @@ solaris-gcc-nas:
 		AUDIO_LIB='-L/usr/openwin/lib -laudio -lXau'\
 		OBJECTS='decode.o dct64.o audio_nas.o' \
 		CFLAGS="$(CFLAGS) $(CPPFLAGS) -O2 -I/usr/openwin/include -Wall \
-			-DSOLARIS -DREAL_IS_FLOAT -DUSE_MMAP \
+			-DSOLARIS -DREAL_IS_FLOAT -DHAVE_MMAP \
 			-DREAD_MMAP -DNAS \
 			-funroll-all-loops -finline-functions" \
 		mpg123-make
@@ -465,7 +465,7 @@ solaris-gcc-nas:
 sunos:
 	$(MAKE) CC=gcc \
 		OBJECTS='decode.o dct64.o audio_sun.o' \
-		CFLAGS="$(CFLAGS) $(CPPFLAGS) -O2 -DSUNOS -DREAL_IS_FLOAT -DUSE_MMAP \
+		CFLAGS="$(CFLAGS) $(CPPFLAGS) -O2 -DSUNOS -DREAL_IS_FLOAT -DHAVE_MMAP \
 			-funroll-loops" \
 		mpg123-make
 
@@ -491,24 +491,24 @@ hpux-gcc:
 sgi:
 	$(MAKE) CC=cc \
 		OBJECTS='decode.o dct64.o audio_sgi.o' AUDIO_LIB=-laudio \
-		CFLAGS="$(CFLAGS) $(CPPFLAGS) -O2 -DSGI -DREAL_IS_FLOAT -DUSE_MMAP' \
+		CFLAGS="$(CFLAGS) $(CPPFLAGS) -O2 -DSGI -DREAL_IS_FLOAT -DHAVE_MMAP' \
 		mpg123-make
 
 sgi-gcc:
 	$(MAKE) CC=gcc \
 		OBJECTS='decode.o dct64.o audio_sgi.o' AUDIO_LIB=-laudio \
-		CFLAGS="$(CFLAGS) $(CPPFLAGS) -O2 -DSGI -DREAL_IS_FLOAT -DUSE_MMAP" \
+		CFLAGS="$(CFLAGS) $(CPPFLAGS) -O2 -DSGI -DREAL_IS_FLOAT -DHAVE_MMAP" \
 		mpg123-make
 
 dec:
 	$(MAKE) CC=cc OBJECTS='decode.o dct64.o audio_dummy.o' \
-		CFLAGS="$(CFLAGS) $(CPPFLAGS) -std1 -warnprotos -O4 -DUSE_MMAP" \
+		CFLAGS="$(CFLAGS) $(CPPFLAGS) -std1 -warnprotos -O4 -DHAVE_MMAP" \
 		LDFLAGS="$(LDFLAGS) -lrt" \
 		mpg123-make
 
 dec-debug:
 	$(MAKE) CC=cc OBJECTS='decode.o dct64.o audio_dummy.o' \
-		CFLAGS="$(CFLAGS) $(CPPFLAGS) -g3 -std1 -warnprotos -O4 -DUSE_MMAP" \
+		CFLAGS="$(CFLAGS) $(CPPFLAGS) -g3 -std1 -warnprotos -O4 -DHAVE_MMAP" \
 		LDFLAGS="$(LDFLAGS) -lrt" \
 		mpg123-make
 
@@ -516,7 +516,7 @@ dec-nas:
 	$(MAKE) CC=cc LDFLAGS="$(LDFLAGS) -L/usr/X11R6/lib" \
 		AUDIO_LIB='-laudio -lXau -ldnet_stub'\
 		OBJECTS='decode.o dct64.o  audio_nas.o' \
-		CFLAGS="$(CFLAGS) $(CPPFLAGS) -I/usr/X11R6/include -std1 -warnprotos -O4 -DUSE_MMAP" \
+		CFLAGS="$(CFLAGS) $(CPPFLAGS) -I/usr/X11R6/include -std1 -warnprotos -O4 -DHAVE_MMAP" \
 		mpg123-make
 
 ultrix:
@@ -526,14 +526,14 @@ ultrix:
 
 aix-gcc:
 	$(MAKE) CC=gcc OBJECTS='decode.o dct64.o audio_aix.o' \
-		CFLAGS="$(CFLAGS) $(CPPFLAGS) -DAIX -Wall -O6 -DUSE_MMAP -DREAD_MMAP -DREAL_IS_FLOAT \
+		CFLAGS="$(CFLAGS) $(CPPFLAGS) -DAIX -Wall -O6 -DHAVE_MMAP -DREAD_MMAP -DREAL_IS_FLOAT \
 			-fomit-frame-pointer -funroll-all-loops \
 			-finline-functions -ffast-math" \
 		mpg123-make
 
 aix-xlc:
 	$(MAKE) OBJECTS='decode.o dct64.o audio_aix.o' \
-		CFLAGS="$(CFLAGS) $(CPPFLAGS) -O3 -qstrict -qcpluscmt -DAIX -DUSE_MMAP \
+		CFLAGS="$(CFLAGS) $(CPPFLAGS) -O3 -qstrict -qcpluscmt -DAIX -DHAVE_MMAP \
 			-DREAD_MMAP" \
 		mpg123-make
 
@@ -543,7 +543,7 @@ aix-tk3play:
 aix-frontend:
 	$(MAKE) OBJECTS='decode.o dct64.o audio_aix.o \
 			control_sajber.o control_tk3play.o' \
-		CFLAGS="$(CFLAGS) $(CPPFLAGS) -DAIX -Wall -O6 -DUSE_MMAP -DFRONTEND \
+		CFLAGS="$(CFLAGS) $(CPPFLAGS) -DAIX -Wall -O6 -DHAVE_MMAP -DFRONTEND \
 			-fomit-frame-pointer -funroll-all-loops \
 			-finline-functions -ffast-math" \
 		$(FRONTEND)
@@ -562,7 +562,7 @@ netbsd:
 		OBJECTS='decode.o dct64.o audio_sun.o' \
 		CFLAGS="$(CFLAGS) $(CPPFLAGS) -Wall -ansi -pedantic -O3 -fomit-frame-pointer \
 			-funroll-all-loops -ffast-math \
-			-DREAL_IS_FLOAT -DUSE_MMAP -DNETBSD -DNO_RT" \
+			-DREAL_IS_FLOAT -DHAVE_MMAP -DNETBSD -DNO_RT" \
 		mpg123-make
 
 netbsd-x86:
@@ -570,7 +570,7 @@ netbsd-x86:
 		OBJECTS='decode_i386.o dct64_i386.o audio_sun.o' \
 		CFLAGS="$(CFLAGS) $(CPPFLAGS) -Wall -ansi -pedantic -O4 -mcpu=i486 -fomit-frame-pointer \
 			-funroll-all-loops -ffast-math -DROT_I386 \
-			-DI386_ASSEM -DREAL_IS_FLOAT -DUSE_MMAP -DNETBSD -DNO_RT" \
+			-DI386_ASSEM -DREAL_IS_FLOAT -DHAVE_MMAP -DNETBSD -DNO_RT" \
 		mpg123-make
 
 netbsd-rt:
@@ -578,7 +578,7 @@ netbsd-rt:
 		OBJECTS='decode.o dct64.o audio_sun.o' \
 		CFLAGS="$(CFLAGS) $(CPPFLAGS) -Wall -ansi -pedantic -O3 -fomit-frame-pointer \
 			-funroll-all-loops -ffast-math \
-			-DREAL_IS_FLOAT -DUSE_MMAP -DNETBSD" \
+			-DREAL_IS_FLOAT -DHAVE_MMAP -DNETBSD" \
 		mpg123-make
 
 netbsd-x86-rt:
@@ -586,7 +586,7 @@ netbsd-x86-rt:
 		OBJECTS='decode_i386.o dct64_i386.o audio_sun.o' \
 		CFLAGS="$(CFLAGS) $(CPPFLAGS) -Wall -ansi -pedantic -O4 -mcpu=i486 -fomit-frame-pointer \
 			-funroll-all-loops -ffast-math -DROT_I386 \
-			-DI386_ASSEM -DREAL_IS_FLOAT -DUSE_MMAP -DNETBSD" \
+			-DI386_ASSEM -DREAL_IS_FLOAT -DHAVE_MMAP -DNETBSD" \
 		mpg123-make
 
 bsdos:
@@ -595,7 +595,7 @@ bsdos:
 			 audio_oss.o' \
 		CFLAGS="$(CFLAGS) $(CPPFLAGS) -Wall -O4 -mcpu=i486 -fomit-frame-pointer \
 			-funroll-all-loops -ffast-math -DROT_I386 \
-			-DI386_ASSEM -DREAL_IS_FLOAT -DUSE_MMAP -DOSS \
+			-DI386_ASSEM -DREAL_IS_FLOAT -DHAVE_MMAP -DOSS \
 			-DDONT_CATCH_SIGNALS" \
 		mpg123-make
 
@@ -604,7 +604,7 @@ bsdos4:
 		OBJECTS='decode_i386.o dct64_i386.o audio_oss.o' \
 		CFLAGS="$(CFLAGS) $(CPPFLAGS) -Wall -O4 -mcpu=i486 -fomit-frame-pointer \
 			-funroll-all-loops -ffast-math -DROT_I386 \
-			-DI386_ASSEM -DREAL_IS_FLOAT -DUSE_MMAP -DOSS \
+			-DI386_ASSEM -DREAL_IS_FLOAT -DHAVE_MMAP -DOSS \
 			-DREAD_MMAP -DDONT_CATCH_SIGNALS" \
 		mpg123-make
 
@@ -615,7 +615,7 @@ bsdos-nas:
 			audio_nas.o' \
 		CFLAGS="$(CFLAGS) $(CPPFLAGS) -Wall -O4 -mcpu=i486 -fomit-frame-pointer \
 			-funroll-all-loops -ffast-math -DROT_I386 \
-			-DI386_ASSEM -DREAL_IS_FLOAT -DUSE_MMAP -DOSS \
+			-DI386_ASSEM -DREAL_IS_FLOAT -DHAVE_MMAP -DOSS \
 			-DDONT_CATCH_SIGNALS -DNAS" \
 		mpg123-make
 
diff --git a/src/audio.c b/src/audio.c
index 9901210a..3cc2d7a7 100644
--- a/src/audio.c
+++ b/src/audio.c
@@ -3,20 +3,12 @@
 
 void audio_info_struct_init(struct audio_info_struct *ai)
 {
-#ifdef AUDIO_USES_FD
   ai->fn = -1;
-#endif
-#ifdef SGI
-#if 0
-  ALconfig config;
-  ALport port;
-#endif
-#endif
   ai->rate = -1;
   ai->gain = -1;
   ai->output = -1;
-#ifdef ALSA
   ai->handle = NULL;
+#ifdef ALSA
   ai->alsa_format.format = -1;
   ai->alsa_format.rate = -1;
   ai->alsa_format.channels = -1;
@@ -26,6 +18,20 @@ void audio_info_struct_init(struct audio_info_struct *ai)
   ai->format = -1;
 }
 
+
+void audio_info_struct_dump(struct audio_info_struct *ai)
+{
+	fprintf(stderr, "ai->fn=%d\n", ai->fn);
+	fprintf(stderr, "ai->handle=%p\n", ai->handle);
+	fprintf(stderr, "ai->rate=%ld\n", ai->rate);
+	fprintf(stderr, "ai->gain=%ld\n", ai->gain);
+	fprintf(stderr, "ai->output=%d\n", ai->output);
+	fprintf(stderr, "ai->device='%s'\n", ai->device);
+	fprintf(stderr, "ai->channels=%d\n", ai->channels);
+	fprintf(stderr, "ai->format=%d\n", ai->format);
+}
+
+
 #define NUM_CHANNELS 2
 #define NUM_ENCODINGS 6
 #define NUM_RATES 10
diff --git a/src/audio.h b/src/audio.h
index a0d535b6..c5a23cfb 100644
--- a/src/audio.h
+++ b/src/audio.h
@@ -6,8 +6,16 @@
 #define AUDIO_OUT_INTERNAL_SPEAKER 0x02
 #define AUDIO_OUT_LINE_OUT         0x04
 
-enum { DECODE_TEST, DECODE_AUDIO, DECODE_FILE, DECODE_BUFFER, DECODE_WAV,
-	DECODE_AU,DECODE_CDR,DECODE_AUDIOFILE };
+enum {
+	DECODE_TEST,
+	DECODE_AUDIO,
+	DECODE_FILE,
+	DECODE_BUFFER,
+	DECODE_WAV,
+	DECODE_AU,
+	DECODE_CDR,
+	DECODE_AUDIOFILE
+};
 
 #define AUDIO_FORMAT_MASK	  0x100
 #define AUDIO_FORMAT_16		  0x100
@@ -23,44 +31,21 @@ enum { DECODE_TEST, DECODE_AUDIO, DECODE_FILE, DECODE_BUFFER, DECODE_WAV,
 /* 3% rate tolerance */
 #define AUDIO_RATE_TOLERANCE	  3
 
-#if 0
-#if defined(HPUX) || defined(SUNOS) || defined(SOLARIS) || defined(OSS) || defined(__NetBSD__) || defined(SPARCLINUX) || defined(__FreeBSD__)
-#endif
-#endif
 
-#define AUDIO_USES_FD
-
-#ifdef SGI
-/* #include <audio.h> */
-#include <dmedia/audio.h>
-#endif
-
-
-#ifdef ALSA
-#include <sys/asoundlib.h>
-#endif
 
 struct audio_info_struct
 {
-#ifdef AUDIO_USES_FD
   int fn; /* filenumber */
-#endif
-#ifdef SGI
-  ALconfig config;
-  ALport port;
-#endif
+  void *handle;	/* driver specific pointer */
+
   long rate;
   long gain;
   int output;
-#ifdef ALSA
-  void *handle;
-  snd_pcm_format_t alsa_format;
-#endif
+
   char *device;
   int channels;
   int format;
-  int private1;
-  void *private2;
+
 };
 
 struct audio_name {
@@ -69,21 +54,21 @@ struct audio_name {
   char *sname;
 };
 
+
+/* ------ Declarations from "audio.c" ------ */
+
+extern void audio_info_struct_init(struct audio_info_struct *);
+extern void audio_info_struct_dump(struct audio_info_struct *ai);
 extern void audio_capabilities(struct audio_info_struct *);
 extern void audio_fit_capabilities(struct audio_info_struct *ai,int c,int r);
-
 extern char *audio_encoding_name(int format);
 
-extern int audio_play_samples(struct audio_info_struct *,unsigned char *,int);
-extern int audio_open(struct audio_info_struct *);
-extern int audio_reset_parameters(struct audio_info_struct *);
-extern int audio_rate_best_match(struct audio_info_struct *ai);
-extern int audio_set_rate(struct audio_info_struct *);
-extern int audio_set_format(struct audio_info_struct *);
-extern int audio_get_formats(struct audio_info_struct *);
-extern int audio_set_channels(struct audio_info_struct *);
-extern int audio_write_sample(struct audio_info_struct *,short *,int);
-extern int audio_close(struct audio_info_struct *);
-extern void audio_info_struct_init(struct audio_info_struct *);
-extern void audio_queueflush(struct audio_info_struct *ai);
+
+/* ------ Declarations from "audio_*.c" ------ */
+
+extern int audio_open(struct audio_info_struct *);
+extern int audio_get_formats(struct audio_info_struct *);
+extern int audio_play_samples(struct audio_info_struct *, unsigned char *,int);
+extern void audio_queueflush(struct audio_info_struct *ai);
+extern int audio_close(struct audio_info_struct *);
 
diff --git a/src/audio_aix.c b/src/audio_aix.c
index a03c8971..38dc88e1 100644
--- a/src/audio_aix.c
+++ b/src/audio_aix.c
@@ -30,6 +30,30 @@
 /* #define AUDIO_BSIZE AUDIO_IGNORE */
 #define AUDIO_BSIZE 200
 
+int audio_rate_best_match(struct audio_info_struct *ai)
+{
+  static long valid [ ] = {  5510,  6620,  8000,  9600, 11025, 16000, 18900,
+                            22050, 27420, 32000, 33075, 37800, 44100, 48000, 0 };
+  int  i = 0;
+  long best = 8000;
+
+  if(!ai || ai->fn < 0 || ai->rate < 0) {
+    return -1;
+  } 
+
+  while (valid [i])
+  {
+    if (abs(valid[i] - ai->rate) < abs(best - ai->rate))
+    {
+      best = valid [i];
+    }
+    i = i + 1;
+  }
+
+  ai->rate = best;
+  return best;
+}
+
 
 int audio_open(struct audio_info_struct *ai)
 {
@@ -182,45 +206,6 @@ int audio_reset_parameters(struct audio_info_struct *ai)
   return 0;
 }
 
-int audio_rate_best_match(struct audio_info_struct *ai)
-{
-  static long valid [ ] = {  5510,  6620,  8000,  9600, 11025, 16000, 18900,
-                            22050, 27420, 32000, 33075, 37800, 44100, 48000, 0 };
-  int  i = 0;
-  long best = 8000;
-
-  if(!ai || ai->fn < 0 || ai->rate < 0) {
-    return -1;
-  } 
-
-  while (valid [i])
-  {
-    if (abs(valid[i] - ai->rate) < abs(best - ai->rate))
-    {
-      best = valid [i];
-    }
-    i = i + 1;
-  }
-
-  ai->rate = best;
-  return best;
-}
-
-int audio_set_rate(struct audio_info_struct *ai)
-{
-  return audio_reset_parameters(ai);
-}
-
-int audio_set_channels(struct audio_info_struct *ai)
-{
-  return audio_reset_parameters(ai);
-}
-
-int audio_set_format(struct audio_info_struct *ai)
-{
-  return audio_reset_parameters(ai);
-}
-
 int audio_get_formats(struct audio_info_struct *ai)
 {
 /* ULTIMEDIA DOCUMENTATION SAYS:
@@ -285,3 +270,8 @@ int audio_close(struct audio_info_struct *ai)
 
     return 0;
 }
+
+void audio_queueflush(struct audio_info_struct *ai)
+{
+}
+
diff --git a/src/audio_alib.c b/src/audio_alib.c
index cd1b631a..e840d9af 100644
--- a/src/audio_alib.c
+++ b/src/audio_alib.c
@@ -175,23 +175,15 @@ inline int audio_play_samples(struct audio_info_struct *ai,unsigned char *buf,in
 
 /**************************************************************************/
 
-/* unused */
-int audio_set_rate(struct audio_info_struct *ai) {
-  return 0;
-}
-
-/**************************************************************************/
-
-/* unused */
-int audio_set_channels(struct audio_info_struct *ai) {
- return 0;
-}
-
 int audio_get_formats(struct audio_info_struct *ai)
 {
   return AUDIO_FORMAT_SIGNED_16;
 }
 
+void audio_queueflush(struct audio_info_struct *ai)
+{
+}
+
 
 /**************************************************************************
  * T H A T ' S    A L L   F O L K S
diff --git a/src/audio_alsa.c b/src/audio_alsa.c
index 38572ed9..f1ef67ae 100644
--- a/src/audio_alsa.c
+++ b/src/audio_alsa.c
@@ -30,6 +30,9 @@
 #define SND_PCM_FMT_U16_NE SND_PCM_FMT_U16_BE
 #endif
 
+
+
+
 int audio_open(struct audio_info_struct *ai)
 {
 	int err;
@@ -38,6 +41,7 @@ int audio_open(struct audio_info_struct *ai)
 
 	if(!ai)
 		return -1;
+		
 	if(ai->device) {	/* parse ALSA device name */
 		if(strchr(ai->device,':')) {	/* card with device */
 			strncpy(scard, ai->device, sizeof(scard)-1);
@@ -60,21 +64,62 @@ int audio_open(struct audio_info_struct *ai)
 		}
 	}
 
+	// Open the ALSA device
 	if((err=snd_pcm_open(&ai->handle, card, device, SND_PCM_OPEN_PLAYBACK)) < 0 )
 	{
 		fprintf(stderr, "open failed: %s\n", snd_strerror(err));
 		exit(1);
 	}
 
-	if(audio_reset_parameters(ai) < 0)
-	{
-		audio_close(ai);
-		return -1;
-	}
+
+	// Now configure the device
+	audio_set_playback_format( ai )
+	audio_set_playback_params( ai );
+
 
 	return 0;
 }
 
+
+static void audio_set_playback_format(struct audio_info_struct *ai)
+{
+	snd_pcm_format_t alsa_format;
+	int err;
+	
+	alsa_format.rat = ai->rate;
+	alsa_format.channels = ai->channels;
+
+	switch(ai->format)
+	{
+		case AUDIO_FORMAT_SIGNED_16:
+		default:
+			ai->alsa_format.format=SND_PCM_SFMT_S16_NE;
+			break;
+		case AUDIO_FORMAT_UNSIGNED_8:
+			ai->alsa_format.format=SND_PCM_SFMT_U8;
+			break;
+		case AUDIO_FORMAT_SIGNED_8:
+			ai->alsa_format.format=SND_PCM_SFMT_S8;
+			break;
+		case AUDIO_FORMAT_ULAW_8:
+			ai->alsa_format.format=SND_PCM_SFMT_MU_LAW;
+			break;
+		case AUDIO_FORMAT_ALAW_8:
+			ai->alsa_format.format=SND_PCM_SFMT_A_LAW;
+			break;
+		case AUDIO_FORMAT_UNSIGNED_16:
+			ai->alsa_format.format=SND_PCM_SFMT_U16_NE;
+			break;
+	}
+
+	if((err=snd_pcm_playback_format(ai->handle, &alsa_format)) < 0 )
+	{
+		fprintf(stderr, "snd_pcm_playback_format failed: %s\n", snd_strerror(err));
+		exit(1);
+	}
+}
+
+
 static void audio_set_playback_params(struct audio_info_struct *ai)
 {
 	int err;
@@ -101,91 +146,8 @@ static void audio_set_playback_params(struct audio_info_struct *ai)
 	}
 }
 
-int audio_reset_parameters(struct audio_info_struct *ai)
-{
-	audio_set_format(ai);
-	audio_set_channels(ai);
-	audio_set_rate(ai);
 
-	return 0;
-}
 
-int audio_rate_best_match(struct audio_info_struct *ai)
-{
-	return 0;
-}
-
-int audio_set_rate(struct audio_info_struct *ai)
-{
-	int ret;
-
-	if(!ai || ai->rate < 0)
-		return -1;
-
-	ai->alsa_format.rate=ai->rate;
-
-	if((ret=snd_pcm_playback_format(ai->handle, &ai->alsa_format)) < 0 )
-		return -1;
-
-	audio_set_playback_params(ai);
-
-	return 0;
-}
-
-int audio_set_channels(struct audio_info_struct *ai)
-{
-	int ret;
-
-	if(ai->alsa_format.channels < 0)
-		return 0;
-
-	ai->alsa_format.channels = ai->channels;
-
-	if((ret=snd_pcm_playback_format(ai->handle, &ai->alsa_format)) < 0 )
-		return -1;
-
-	audio_set_playback_params(ai);
-
-	return 0;
-}
-
-int audio_set_format(struct audio_info_struct *ai)
-{
-	int ret;
-
-	if(ai->format == -1)
-		return 0;
-
-	switch(ai->format)
-	{
-		case AUDIO_FORMAT_SIGNED_16:
-		default:
-			ai->alsa_format.format=SND_PCM_SFMT_S16_NE;
-			break;
-		case AUDIO_FORMAT_UNSIGNED_8:
-			ai->alsa_format.format=SND_PCM_SFMT_U8;
-			break;
-		case AUDIO_FORMAT_SIGNED_8:
-			ai->alsa_format.format=SND_PCM_SFMT_S8;
-			break;
-		case AUDIO_FORMAT_ULAW_8:
-			ai->alsa_format.format=SND_PCM_SFMT_MU_LAW;
-			break;
-		case AUDIO_FORMAT_ALAW_8:
-			ai->alsa_format.format=SND_PCM_SFMT_A_LAW;
-			break;
-		case AUDIO_FORMAT_UNSIGNED_16:
-			ai->alsa_format.format=SND_PCM_SFMT_U16_NE;
-			break;
-	}
-
-	if((ret=snd_pcm_playback_format(ai->handle, &ai->alsa_format)) < 0 )
-		return -1;
-
-	audio_set_playback_params(ai);
-
-	return 0;
-}
 
 int audio_get_formats(struct audio_info_struct *ai)
 {
@@ -236,3 +198,8 @@ int audio_close(struct audio_info_struct *ai)
 	ret = snd_pcm_close(ai->handle);
 	return ret;
 }
+
+void audio_queueflush(struct audio_info_struct *ai)
+{
+}
+
diff --git a/src/audio_dummy.c b/src/audio_dummy.c
index a81ea6eb..0b61e5ea 100644
--- a/src/audio_dummy.c
+++ b/src/audio_dummy.c
@@ -7,31 +7,6 @@ int audio_open(struct audio_info_struct *ai)
   return -1;
 }
 
-int audio_reset_parameters(struct audio_info_struct *ai)
-{
-  return 0;
-}
-
-int audio_rate_best_match(struct audio_info_struct *ai)
-{
-  return 0;
-}
-
-int audio_set_rate(struct audio_info_struct *ai)
-{
-  return 0;
-}
-
-int audio_set_channels(struct audio_info_struct *ai)
-{
-  return 0;
-}
-
-int audio_set_format(struct audio_info_struct *ai)
-{
-  return 0;
-}
-
 int audio_get_formats(struct audio_info_struct *ai)
 {
   return AUDIO_FORMAT_SIGNED_16;
@@ -46,3 +21,7 @@ int audio_close(struct audio_info_struct *ai)
 {
   return 0;
 }
+
+void audio_queueflush(struct audio_info_struct *ai)
+{
+}
diff --git a/src/audio_esd.c b/src/audio_esd.c
index ff8f71c2..3cc780ef 100644
--- a/src/audio_esd.c
+++ b/src/audio_esd.c
@@ -76,38 +76,10 @@ int audio_open(struct audio_info_struct *ai)
   else if (ai->rate > esd_rate)
     return -1;
 
-  ai->fn = esd_play_stream_fallback(format, ai->rate, esdserver, "mpg123");
+  ai->fn = esd_play_stream_fallback(format, ai->rate, ai->device, "mpg123");
   return (ai->fn);
 }
 
-int audio_reset_parameters(struct audio_info_struct *ai)
-{
-  return 0;
-}
-
-int audio_rate_best_match(struct audio_info_struct *ai)
-{
-  return 0;
-}
-
-int audio_set_rate(struct audio_info_struct *ai)
-{
-  audio_close(ai);
-  return audio_open(ai);
-}
-
-int audio_set_channels(struct audio_info_struct *ai)
-{
-  audio_close(ai);
-  return audio_open(ai);
-}
-
-int audio_set_format(struct audio_info_struct *ai)
-{
-  audio_close(ai);
-  return audio_open(ai);
-}
-
 int audio_get_formats(struct audio_info_struct *ai)
 {
   if (0 < ai->channels && ai->channels <= esd_channels
diff --git a/src/audio_hp.c b/src/audio_hp.c
index f99ff13b..c6d12d2a 100644
--- a/src/audio_hp.c
+++ b/src/audio_hp.c
@@ -8,6 +8,65 @@
 
 #include <sys/audio.h>
 
+
+
+static int audio_set_rate(struct audio_info_struct *ai)
+{
+  if(ai->rate >= 0)
+    return ioctl(ai->fn,AUDIO_SET_SAMPLE_RATE,ai->rate);
+  return 0;
+}
+
+static int audio_set_channels(struct audio_info_struct *ai)
+{
+  if(ai->channels<0)
+    return 0;
+  return ioctl(ai->fn,AUDIO_SET_CHANNELS,ai->channels);
+}
+
+static int audio_set_format(struct audio_info_struct *ai)
+{
+  int fmt;
+
+  switch(ai->format) {
+    case -1:
+    case AUDIO_FORMAT_SIGNED_16:
+    default: 
+      fmt = AUDIO_FORMAT_LINEAR16BIT;
+      break;
+    case AUDIO_FORMAT_UNSIGNED_8:
+      fprintf(stderr,"unsigned 8 bit linear not supported\n");
+      return -1;
+    case AUDIO_FORMAT_SIGNED_8:
+      fprintf(stderr,"signed 8 bit linear not supported\n");
+      return -1;
+    case AUDIO_FORMAT_ALAW_8:
+      fmt = AUDIO_FORMAT_ALAW;
+      break;
+    case AUDIO_FORMAT_ULAW_8:
+      fmt = AUDIO_FORMAT_ULAW;
+      break;
+  }
+  return ioctl(ai->fn,AUDIO_SET_DATA_FORMAT,fmt);
+}
+
+static int audio_get_formats(struct audio_info_struct *ai)
+{
+  return AUDIO_FORMAT_SIGNED_16;
+}
+
+static int audio_reset_parameters(struct audio_info_struct *ai)
+{
+  int ret;
+  ret = audio_set_format(ai);
+  if(ret >= 0)
+    ret = audio_set_channels(ai);
+  if(ret >= 0)
+    ret = audio_set_rate(ai);
+  return ret;
+}
+
+
 int audio_open(struct audio_info_struct *ai)
 {
   struct audio_describe ades;
@@ -72,66 +131,6 @@ int audio_open(struct audio_info_struct *ai)
   return ai->fn;
 }
 
-int audio_reset_parameters(struct audio_info_struct *ai)
-{
-  int ret;
-  ret = audio_set_format(ai);
-  if(ret >= 0)
-    ret = audio_set_channels(ai);
-  if(ret >= 0)
-    ret = audio_set_rate(ai);
-  return ret;
-}
-
-int audio_rate_best_match(struct audio_info_struct *ai)
-{
-  return 0;
-}
-
-int audio_set_rate(struct audio_info_struct *ai)
-{
-  if(ai->rate >= 0)
-    return ioctl(ai->fn,AUDIO_SET_SAMPLE_RATE,ai->rate);
-  return 0;
-}
-
-int audio_set_channels(struct audio_info_struct *ai)
-{
-  if(ai->channels<0)
-    return 0;
-  return ioctl(ai->fn,AUDIO_SET_CHANNELS,ai->channels);
-}
-
-int audio_set_format(struct audio_info_struct *ai)
-{
-  int fmt;
-
-  switch(ai->format) {
-    case -1:
-    case AUDIO_FORMAT_SIGNED_16:
-    default: 
-      fmt = AUDIO_FORMAT_LINEAR16BIT;
-      break;
-    case AUDIO_FORMAT_UNSIGNED_8:
-      fprintf(stderr,"unsigned 8 bit linear not supported\n");
-      return -1;
-    case AUDIO_FORMAT_SIGNED_8:
-      fprintf(stderr,"signed 8 bit linear not supported\n");
-      return -1;
-    case AUDIO_FORMAT_ALAW_8:
-      fmt = AUDIO_FORMAT_ALAW;
-      break;
-    case AUDIO_FORMAT_ULAW_8:
-      fmt = AUDIO_FORMAT_ULAW;
-      break;
-  }
-  return ioctl(ai->fn,AUDIO_SET_DATA_FORMAT,fmt);
-}
-
-int audio_get_formats(struct audio_info_struct *ai)
-{
-  return AUDIO_FORMAT_SIGNED_16;
-}
 
 
 int audio_play_samples(struct audio_info_struct *ai,unsigned char *buf,int len)
@@ -144,3 +143,8 @@ int audio_close(struct audio_info_struct *ai)
   close (ai->fn);
   return 0;
 }
+
+void audio_queueflush(struct audio_info_struct *ai)
+{
+}
+
diff --git a/src/audio_libao.c b/src/audio_libao.c
index bf610559..3586ed62 100644
--- a/src/audio_libao.c
+++ b/src/audio_libao.c
@@ -123,7 +123,7 @@ int audio_play_samples(struct audio_info_struct *ai,unsigned char *buf,int len)
 	int res = 0;
 	ao_device *device = (ao_device*)ai->handle;
 	
-	res = ao_play(device, buf, len);
+	res = ao_play(device, (char*)buf, len);
 	if (res==0) {
 		fprintf(stderr, "audio_play_samples(): error playing samples\n");
 		return -1;
diff --git a/src/audio_macosx.c b/src/audio_macosx.c
index 707e82a8..8fa57b94 100644
--- a/src/audio_macosx.c
+++ b/src/audio_macosx.c
@@ -223,7 +223,7 @@ void start(int n)
 
 int audio_open(struct audio_info_struct *ai)
 {
-	long size;
+	UInt32 size;
 	AudioStreamBasicDescription format;
 	#if MOSX_USES_SEM
 	char s[10];
diff --git a/src/audio_mint.c b/src/audio_mint.c
index dc0a911d..94cae044 100644
--- a/src/audio_mint.c
+++ b/src/audio_mint.c
@@ -8,6 +8,93 @@ extern int outburst;
 
 int real_rate_printed = 0;
 
+
+
+static int audio_rate_best_match(struct audio_info_struct *ai)
+{
+  int ret,dsp_rate;
+
+  if(!ai || ai->fn < 0 || ai->rate < 0)
+    return -1;
+  dsp_rate = ai->rate;
+  ret = ioctl(ai->fn,AIOCSSPEED, (void *)dsp_rate);
+  ret = ioctl(ai->fn,AIOCGSPEED,&dsp_rate);
+  if(ret < 0)
+    return ret;
+  ai->rate = dsp_rate;
+  return 0;
+}
+
+static int audio_set_rate(struct audio_info_struct *ai)
+{
+  int dsp_rate = ai->rate;
+
+  if(ai->rate >= 0) {
+    int ret, real_rate;
+    ret = ioctl(ai->fn, AIOCSSPEED, (void *)dsp_rate);
+    if (ret >= 0 && !real_rate_printed) {
+      ioctl(ai->fn,AIOCGSPEED,&real_rate);
+      if (real_rate != dsp_rate) {
+        fprintf(stderr, "Replay rate: %d Hz\n", real_rate);
+        real_rate_printed = 1;
+      }
+    }
+    return ret;
+  }
+
+  return 0;
+}
+
+static int audio_set_channels(struct audio_info_struct *ai)
+{
+  int chan = ai->channels;
+
+  if(ai->channels < 1)
+    return 0;
+
+  return ioctl(ai->fn, AIOCSCHAN, (void *)chan);
+}
+
+static int audio_set_format(struct audio_info_struct *ai)
+{
+  int fmts;
+
+  if(ai->format == -1)
+    return 0;
+
+  switch(ai->format) {
+    case AUDIO_FORMAT_SIGNED_16:
+    default:
+      fmts = AFMT_S16;
+      break;
+    case AUDIO_FORMAT_UNSIGNED_8:
+      fmts = AFMT_U8;
+      break;
+    case AUDIO_FORMAT_SIGNED_8:
+      fmts = AFMT_S8;
+      break;
+    case AUDIO_FORMAT_ULAW_8:
+      fmts = AFMT_ULAW;
+      break;
+  }
+  return ioctl(ai->fn, AIOCSFMT, (void *)fmts);
+}
+
+static int audio_reset_parameters(struct audio_info_struct *ai)
+{
+  int ret;
+  ret = ioctl(ai->fn,AIOCRESET,NULL);
+  if(ret >= 0)
+    ret = audio_set_format(ai);
+  if(ret >= 0)
+    ret = audio_set_channels(ai);
+  if(ret >= 0)
+    ret = audio_set_rate(ai);
+  return ret;
+}
+
+
+
 int audio_open(struct audio_info_struct *ai)
 {
   if(!ai)
@@ -32,90 +119,6 @@ int audio_open(struct audio_info_struct *ai)
   }
   return ai->fn;
 }
-
-int audio_reset_parameters(struct audio_info_struct *ai)
-{
-  int ret;
-  ret = ioctl(ai->fn,AIOCRESET,NULL);
-  if(ret >= 0)
-    ret = audio_set_format(ai);
-  if(ret >= 0)
-    ret = audio_set_channels(ai);
-  if(ret >= 0)
-    ret = audio_set_rate(ai);
-  return ret;
-}
-
-int audio_rate_best_match(struct audio_info_struct *ai)
-{
-  int ret,dsp_rate;
-
-  if(!ai || ai->fn < 0 || ai->rate < 0)
-    return -1;
-  dsp_rate = ai->rate;
-  ret = ioctl(ai->fn,AIOCSSPEED, (void *)dsp_rate);
-  ret = ioctl(ai->fn,AIOCGSPEED,&dsp_rate);
-  if(ret < 0)
-    return ret;
-  ai->rate = dsp_rate;
-  return 0;
-}
-
-int audio_set_rate(struct audio_info_struct *ai)
-{
-  int dsp_rate = ai->rate;
-
-  if(ai->rate >= 0) {
-    int ret, real_rate;
-    ret = ioctl(ai->fn, AIOCSSPEED, (void *)dsp_rate);
-    if (ret >= 0 && !real_rate_printed) {
-      ioctl(ai->fn,AIOCGSPEED,&real_rate);
-      if (real_rate != dsp_rate) {
-        fprintf(stderr, "Replay rate: %d Hz\n", real_rate);
-        real_rate_printed = 1;
-      }
-    }
-    return ret;
-  }
-
-  return 0;
-}
-
-int audio_set_channels(struct audio_info_struct *ai)
-{
-  int chan = ai->channels;
-
-  if(ai->channels < 1)
-    return 0;
-
-  return ioctl(ai->fn, AIOCSCHAN, (void *)chan);
-}
-
-int audio_set_format(struct audio_info_struct *ai)
-{
-  int fmts;
-
-  if(ai->format == -1)
-    return 0;
-
-  switch(ai->format) {
-    case AUDIO_FORMAT_SIGNED_16:
-    default:
-      fmts = AFMT_S16;
-      break;
-    case AUDIO_FORMAT_UNSIGNED_8:
-      fmts = AFMT_U8;
-      break;
-    case AUDIO_FORMAT_SIGNED_8:
-      fmts = AFMT_S8;
-      break;
-    case AUDIO_FORMAT_ULAW_8:
-      fmts = AFMT_ULAW;
-      break;
-  }
-  return ioctl(ai->fn, AIOCSFMT, (void *)fmts);
-}
-
 int audio_get_formats(struct audio_info_struct *ai)
 {
   int ret = 0;
@@ -146,3 +149,8 @@ int audio_close(struct audio_info_struct *ai)
   close (ai->fn);
   return 0;
 }
+
+void audio_queueflush(struct audio_info_struct *ai)
+{
+}
+
diff --git a/src/audio_nas.c b/src/audio_nas.c
index 3bf24e28..1ef6d46e 100644
--- a/src/audio_nas.c
+++ b/src/audio_nas.c
@@ -185,7 +185,7 @@ void nas_createFlow(struct audio_info_struct *ai)
     info.buf_size = buf_samples * ai->channels * AuSizeofFormat(format);
     info.buf = (char *) malloc(info.buf_size);
     if (info.buf == NULL) {
-        fprintf(stderr, "Unable to allocate input/output buffer of size %d\n",
+        fprintf(stderr, "Unable to allocate input/output buffer of size %ld\n",
              info.buf_size);
         exit(1);
     }
@@ -230,43 +230,7 @@ int audio_open(struct audio_info_struct *ai)
     return 0;
 }
     
-int audio_reset_parameters(struct audio_info_struct *ai)
-{
-    int ret;
 
-    ret = audio_close(ai);
-    if (ret >= 0)
-        ret = audio_open(ai);
-    return ret;
-}
-
-extern int audio_rate_best_match(struct audio_info_struct *ai)
-{
-    int maxRate, minRate;
-
-    if(!ai || ai->rate < 0)
-        return -1;
-    maxRate =  AuServerMaxSampleRate(info.aud);
-    minRate =  AuServerMinSampleRate(info.aud);
-    if (ai->rate > maxRate) ai->rate = maxRate;
-    if (ai->rate < minRate) ai->rate = minRate;
-    return 0;
-}
-
-int audio_set_rate(struct audio_info_struct *ai)
-{
-    return 0;
-}
-
-int audio_set_channels(struct audio_info_struct *ai)
-{
-    return 0;
-}
-
-int audio_set_format(struct audio_info_struct *ai)
-{
-    return 0;
-}
 
 int audio_get_formats(struct audio_info_struct *ai)
 {
@@ -337,3 +301,7 @@ int audio_close(struct audio_info_struct *ai)
     
     return 0;
 }
+
+void audio_queueflush(struct audio_info_struct *ai)
+{
+}
diff --git a/src/audio_os2.c b/src/audio_os2.c
index 7d072fca..b8917864 100644
--- a/src/audio_os2.c
+++ b/src/audio_os2.c
@@ -575,11 +575,6 @@ int audio_get_formats(struct audio_info_struct *ai)
    return fmts;
 }
 
-int audio_rate_best_match(struct audio_info_struct *ai)
-{
-   return 0;
-}
-
 int audio_get_devices(char *info, int deviceid)
 {
    char buffer[128];
@@ -632,3 +627,6 @@ int audio_get_devices(char *info, int deviceid)
    }
 }
 
+void audio_queueflush(struct audio_info_struct *ai)
+{
+}
diff --git a/src/audio_oss.c b/src/audio_oss.c
index 9b92af2a..872bb9dc 100644
--- a/src/audio_oss.c
+++ b/src/audio_oss.c
@@ -34,6 +34,124 @@
 
 extern int outburst;
 
+
+static int audio_rate_best_match(struct audio_info_struct *ai)
+{
+  int ret,dsp_rate;
+
+  if(!ai || ai->fn < 0 || ai->rate < 0)
+    return -1;
+  dsp_rate = ai->rate;
+  ret = ioctl(ai->fn, SNDCTL_DSP_SPEED,&dsp_rate);
+  if(ret < 0)
+    return ret;
+  ai->rate = dsp_rate;
+  return 0;
+}
+
+static int audio_set_rate(struct audio_info_struct *ai)
+{
+  int dsp_rate;
+  int ret = 0;
+
+  if(ai->rate >= 0) {
+    dsp_rate = ai->rate;
+    ret = ioctl(ai->fn, SNDCTL_DSP_SPEED,&dsp_rate);
+  }
+  return ret;
+}
+
+static int audio_set_channels(struct audio_info_struct *ai)
+{
+  int chan = ai->channels - 1;
+  int ret;
+
+  if(ai->channels < 0)
+    return 0;
+
+  ret = ioctl(ai->fn, SNDCTL_DSP_STEREO, &chan);
+  if(chan != (ai->channels-1)) {
+    return -1;
+  }
+  return ret;
+}
+
+static int audio_set_format(struct audio_info_struct *ai)
+{
+  int sample_size,fmts;
+  int sf,ret;
+
+  if(ai->format == -1)
+    return 0;
+
+  switch(ai->format) {
+    case AUDIO_FORMAT_SIGNED_16:
+    default:
+      fmts = AFMT_S16_NE;
+      sample_size = 16;
+      break;
+    case AUDIO_FORMAT_UNSIGNED_8:
+      fmts = AFMT_U8;
+      sample_size = 8;
+      break;
+    case AUDIO_FORMAT_SIGNED_8:
+      fmts = AFMT_S8;
+      sample_size = 8;
+      break;
+    case AUDIO_FORMAT_ULAW_8:
+      fmts = AFMT_MU_LAW;
+      sample_size = 8;
+      break;
+    case AUDIO_FORMAT_ALAW_8:
+      fmts = AFMT_A_LAW;
+      sample_size = 8;
+      break;
+    case AUDIO_FORMAT_UNSIGNED_16:
+      fmts = AFMT_U16_NE;
+      break;
+  }
+#if 0
+  if(ioctl(ai->fn, SNDCTL_DSP_SAMPLESIZE,&sample_size) < 0)
+    return -1;
+#endif
+  sf = fmts;
+  ret = ioctl(ai->fn, SNDCTL_DSP_SETFMT, &fmts);
+  if(sf != fmts) {
+    return -1;
+  }
+  return ret;
+}
+
+
+static int audio_reset_parameters(struct audio_info_struct *ai)
+{
+  int ret;
+  ret = ioctl(ai->fn, SNDCTL_DSP_RESET, NULL);
+  if(ret < 0)
+    fprintf(stderr,"Can't reset audio!\n");
+  ret = audio_set_format(ai);
+  if (ret == -1)
+    goto err;
+  ret = audio_set_channels(ai);
+  if (ret == -1)
+    goto err;
+  ret = audio_set_rate(ai);
+  if (ret == -1)
+    goto err;
+
+  /* Careful here.  As per OSS v1.1, the next ioctl() commits the format
+   * set above, so we must issue SNDCTL_DSP_RESET before we're allowed to
+   * change it again. [dk]
+   */
+  if (ioctl(ai->fn, SNDCTL_DSP_GETBLKSIZE, &outburst) == -1 ||
+      outburst > MAXOUTBURST)
+    outburst = MAXOUTBURST;
+
+err:
+  return ret;
+}
+
+
 int audio_open(struct audio_info_struct *ai)
 {
   char usingdefdev = 0;
@@ -90,141 +208,7 @@ int audio_open(struct audio_info_struct *ai)
   return ai->fn;
 }
 
-int audio_reset_parameters(struct audio_info_struct *ai)
-{
-  int ret;
-  ret = ioctl(ai->fn, SNDCTL_DSP_RESET, NULL);
-  if(ret < 0)
-    fprintf(stderr,"Can't reset audio!\n");
-  ret = audio_set_format(ai);
-  if (ret == -1)
-    goto err;
-  ret = audio_set_channels(ai);
-  if (ret == -1)
-    goto err;
-  ret = audio_set_rate(ai);
-  if (ret == -1)
-    goto err;
 
-  /* Careful here.  As per OSS v1.1, the next ioctl() commits the format
-   * set above, so we must issue SNDCTL_DSP_RESET before we're allowed to
-   * change it again. [dk]
-   */
-  if (ioctl(ai->fn, SNDCTL_DSP_GETBLKSIZE, &outburst) == -1 ||
-      outburst > MAXOUTBURST)
-    outburst = MAXOUTBURST;
-
-err:
-  return ret;
-}
-
-static int audio_get_parameters(struct audio_info_struct *ai)
-{
-	int c=-1;
-	int r=-1;
-	int f=-1;
-
-	if(ioctl(ai->fn,SNDCTL_DSP_SPEED,&r) < 0)
-		return -1;
-	if(ioctl(ai->fn,SNDCTL_DSP_STEREO,&c) < 0)
-		return -1;
-	if(ioctl(ai->fn,SNDCTL_DSP_SETFMT,&f) < 0)
-		return -1;
-
-	ai->rate = r;
-	ai->channels = c + 1;
-	ai->format = f;
-
-	return 0;
-}
-
-
-int audio_rate_best_match(struct audio_info_struct *ai)
-{
-  int ret,dsp_rate;
-
-  if(!ai || ai->fn < 0 || ai->rate < 0)
-    return -1;
-  dsp_rate = ai->rate;
-  ret = ioctl(ai->fn, SNDCTL_DSP_SPEED,&dsp_rate);
-  if(ret < 0)
-    return ret;
-  ai->rate = dsp_rate;
-  return 0;
-}
-
-int audio_set_rate(struct audio_info_struct *ai)
-{
-  int dsp_rate;
-  int ret = 0;
-
-  if(ai->rate >= 0) {
-    dsp_rate = ai->rate;
-    ret = ioctl(ai->fn, SNDCTL_DSP_SPEED,&dsp_rate);
-  }
-  return ret;
-}
-
-int audio_set_channels(struct audio_info_struct *ai)
-{
-  int chan = ai->channels - 1;
-  int ret;
-
-  if(ai->channels < 0)
-    return 0;
-
-  ret = ioctl(ai->fn, SNDCTL_DSP_STEREO, &chan);
-  if(chan != (ai->channels-1)) {
-    return -1;
-  }
-  return ret;
-}
-
-int audio_set_format(struct audio_info_struct *ai)
-{
-  int sample_size,fmts;
-  int sf,ret;
-
-  if(ai->format == -1)
-    return 0;
-
-  switch(ai->format) {
-    case AUDIO_FORMAT_SIGNED_16:
-    default:
-      fmts = AFMT_S16_NE;
-      sample_size = 16;
-      break;
-    case AUDIO_FORMAT_UNSIGNED_8:
-      fmts = AFMT_U8;
-      sample_size = 8;
-      break;
-    case AUDIO_FORMAT_SIGNED_8:
-      fmts = AFMT_S8;
-      sample_size = 8;
-      break;
-    case AUDIO_FORMAT_ULAW_8:
-      fmts = AFMT_MU_LAW;
-      sample_size = 8;
-      break;
-    case AUDIO_FORMAT_ALAW_8:
-      fmts = AFMT_A_LAW;
-      sample_size = 8;
-      break;
-    case AUDIO_FORMAT_UNSIGNED_16:
-      fmts = AFMT_U16_NE;
-      break;
-  }
-#if 0
-  if(ioctl(ai->fn, SNDCTL_DSP_SAMPLESIZE,&sample_size) < 0)
-    return -1;
-#endif
-  sf = fmts;
-  ret = ioctl(ai->fn, SNDCTL_DSP_SETFMT, &fmts);
-  if(sf != fmts) {
-    return -1;
-  }
-  return ret;
-}
 
 /*
  * get formats for specific channel/rate parameters
@@ -288,7 +272,7 @@ fprintf(stderr,"No");
 
 int audio_play_samples(struct audio_info_struct *ai,unsigned char *buf,int len)
 {
-#ifdef PPC_ENDIAN
+#ifdef WORDS_BIGENDIAN
 #define BYTE0(n) ((unsigned char)(n) & (0xFF))
 #define BYTE1(n) BYTE0((unsigned int)(n) >> 8)
 #define BYTE2(n) BYTE0((unsigned int)(n) >> 16)
@@ -311,7 +295,7 @@ int audio_play_samples(struct audio_info_struct *ai,unsigned char *buf,int len)
          intPtr++;
        }
     }
-#endif /* PPC_ENDIAN */
+#endif /* WORDS_BIGENDIAN */
 
   return write(ai->fn,buf,len);
 }
@@ -321,3 +305,7 @@ int audio_close(struct audio_info_struct *ai)
   close (ai->fn);
   return 0;
 }
+
+void audio_queueflush(struct audio_info_struct *ai)
+{
+}
diff --git a/src/audio_sgi.c b/src/audio_sgi.c
index 00ac8164..d269ae71 100644
--- a/src/audio_sgi.c
+++ b/src/audio_sgi.c
@@ -4,6 +4,9 @@
 #include <fcntl.h>
 #include <stdlib.h>
 
+/* #include <audio.h> */
+#include <dmedia/audio.h>
+
 #include "mpg123.h"
 
 
@@ -11,27 +14,75 @@
 static const char analog_output_res_name[] = ".AnalogOut";
 
 
+static int audio_set_rate(struct audio_info_struct *ai, ALconfig config)
+{
+  int dev = alGetDevice(config);
+  ALpv params[1];
+  
+  /* Make sure the device is OK */
+  if (dev < 0)
+    {
+      fprintf(stderr,"audio_set_rate : %s\n",alGetErrorString(oserror()));
+      return 1;      
+    }
+
+  params[0].param = AL_OUTPUT_RATE;
+  params[0].value.ll = alDoubleToFixed(ai->rate);
+  
+  if (alSetParams(dev, params,1) < 0)
+    fprintf(stderr,"audio_set_rate : %s\n",alGetErrorString(oserror()));
+  
+  return 0;
+}
+
+static int audio_set_channels(struct audio_info_struct *ai, ALconfig config)
+{
+  int ret;
+  
+  if(ai->channels == 2)
+    ret = alSetChannels(config, AL_STEREO);
+  else
+    ret = alSetChannels(config, AL_MONO);
+
+  if (ret < 0)
+    fprintf(stderr,"audio_set_channels : %s\n",alGetErrorString(oserror()));
+  
+  return 0;
+}
+
+static int audio_set_format(struct audio_info_struct *ai, ALconfig config)
+{
+  if (alSetSampFmt(config,AL_SAMPFMT_TWOSCOMP) < 0)
+    fprintf(stderr,"audio_set_format : %s\n",alGetErrorString(oserror()));
+  
+  if (alSetWidth(config,AL_SAMPLE_16) < 0)
+    fprintf(stderr,"audio_set_format : %s\n",alGetErrorString(oserror()));
+  
+  return 0;
+}
+
+
 int audio_open(struct audio_info_struct *ai)
 {
   int dev = AL_DEFAULT_OUTPUT;
-
-  ai->config = alNewConfig();
-
+  ALconfig config = alNewConfig();
+  ALport port = NULL;
+  
   /* Test for correct completion */
-  if (ai->config == 0) {
+  if (config == 0) {
     fprintf(stderr,"audio_open : %s\n",alGetErrorString(oserror()));
     exit(-1);
   }
   
   /* Set port parameters */
   if(ai->channels == 2)
-    alSetChannels(ai->config, AL_STEREO);
+    alSetChannels(config, AL_STEREO);
   else
-    alSetChannels(ai->config, AL_MONO);
+    alSetChannels(config, AL_MONO);
 
-  alSetWidth(ai->config, AL_SAMPLE_16);
-  alSetSampFmt(ai->config,AL_SAMPFMT_TWOSCOMP);
-  alSetQueueSize(ai->config, 131069);
+  alSetWidth(config, AL_SAMPLE_16);
+  alSetSampFmt(config,AL_SAMPFMT_TWOSCOMP);
+  alSetQueueSize(config, 131069);
 
   /* Setup output device to specified module. If there is no module
      specified in ai structure, use the default four output */
@@ -59,90 +110,33 @@ int audio_open(struct audio_info_struct *ai)
   }
   
   /* Set the device */
-  if (alSetDevice(ai->config,dev) < 0)
+  if (alSetDevice(config,dev) < 0)
     {
       fprintf(stderr,"audio_open : %s\n",alGetErrorString(oserror()));
       exit(-1);
     }
   
   /* Open the audio port */
-  ai->port = alOpenPort("mpg123-VSC", "w", ai->config);
-  if(ai->port == NULL) {
+  port = alOpenPort("mpg123-VSC", "w", config);
+  if(port == NULL) {
     fprintf(stderr, "Unable to open audio channel: %s\n",
           alGetErrorString(oserror()));
     exit(-1);
   }
   
-  audio_reset_parameters(ai);
+  ai->handle = (void*)port;
+  
+  
+  audio_set_format(ai, config);
+  audio_set_channels(ai, config);
+  audio_set_rate(ai, config);
     
+
+  alFreeConfig(config);
+ 
   return 1;
 }
 
-int audio_reset_parameters(struct audio_info_struct *ai)
-{
-  int ret;
-  ret = audio_set_format(ai);
-  if(ret >= 0)
-    ret = audio_set_channels(ai);
-  if(ret >= 0)
-    ret = audio_set_rate(ai);
-
-/* todo: Set new parameters here */
-
-  return ret;
-}
-
-int audio_rate_best_match(struct audio_info_struct *ai)
-{
-  return 0;
-}
-
-int audio_set_rate(struct audio_info_struct *ai)
-{
-  int dev = alGetDevice(ai->config);
-  ALpv params[1];
-  
-  /* Make sure the device is OK */
-  if (dev < 0)
-    {
-      fprintf(stderr,"audio_set_rate : %s\n",alGetErrorString(oserror()));
-      return 1;      
-    }
-
-  params[0].param = AL_OUTPUT_RATE;
-  params[0].value.ll = alDoubleToFixed(ai->rate);
-  
-  if (alSetParams(dev, params,1) < 0)
-    fprintf(stderr,"audio_set_rate : %s\n",alGetErrorString(oserror()));
-  
-  return 0;
-}
-
-int audio_set_channels(struct audio_info_struct *ai)
-{
-  int ret;
-  
-  if(ai->channels == 2)
-    ret = alSetChannels(ai->config, AL_STEREO);
-  else
-    ret = alSetChannels(ai->config, AL_MONO);
-
-  if (ret < 0)
-    fprintf(stderr,"audio_set_channels : %s\n",alGetErrorString(oserror()));
-  
-  return 0;
-}
-
-int audio_set_format(struct audio_info_struct *ai)
-{
-  if (alSetSampFmt(ai->config,AL_SAMPFMT_TWOSCOMP) < 0)
-    fprintf(stderr,"audio_set_format : %s\n",alGetErrorString(oserror()));
-  
-  if (alSetWidth(ai->config,AL_SAMPLE_16) < 0)
-    fprintf(stderr,"audio_set_format : %s\n",alGetErrorString(oserror()));
-  
-  return 0;
-}
 
 int audio_get_formats(struct audio_info_struct *ai)
 {
@@ -152,22 +146,30 @@ int audio_get_formats(struct audio_info_struct *ai)
 
 int audio_play_samples(struct audio_info_struct *ai,unsigned char *buf,int len)
 {
+  ALport port = (ALport)ai->handle;
+
   if(ai->format == AUDIO_FORMAT_SIGNED_8)
-    alWriteFrames(ai->port, buf, len>>1);
+    alWriteFrames(port, buf, len>>1);
   else
-    alWriteFrames(ai->port, buf, len>>2);
+    alWriteFrames(port, buf, len>>2);
 
   return len;
 }
 
 int audio_close(struct audio_info_struct *ai)
 {
-  if (ai->port) {
-    while(alGetFilled(ai->port) > 0)
+  ALport port = (ALport)ai->handle;
+
+  if (port) {
+    while(alGetFilled(port) > 0)
       sginap(1);  
-    alClosePort(ai->port);
-    alFreeConfig(ai->config);
+    alClosePort(port);
+    ai->handle=NULL;
   }
   
   return 0;
 }
+
+void audio_queueflush(struct audio_info_struct *ai)
+{
+}
diff --git a/src/audio_sun.c b/src/audio_sun.c
index d3940ba3..54a427fe 100644
--- a/src/audio_sun.c
+++ b/src/audio_sun.c
@@ -29,8 +29,114 @@
 #include <sys/ioctl.h>
 #endif
 
-static void audio_set_format_helper(struct audio_info_struct *ai,audio_info_t *ainfo);
 
+static int audio_reset_parameters(struct audio_info_struct *ai)
+{
+  audio_info_t ainfo;
+
+  AUDIO_INITINFO(&ainfo);
+
+  if(ai->rate != -1)
+    ainfo.play.sample_rate = ai->rate;
+  if(ai->channels >= 0)
+    ainfo.play.channels = ai->channels;
+  audio_set_format_helper(ai,&ainfo);
+
+  if(ioctl(ai->fn, AUDIO_SETINFO, &ainfo) == -1)
+    return -1;
+  return 0;
+}
+
+static int audio_rate_best_match(struct audio_info_struct *ai)
+{
+  audio_info_t ainfo;
+  AUDIO_INITINFO(&ainfo);
+ 
+  ainfo.play.sample_rate = ai->rate;
+  if(ioctl(ai->fn, AUDIO_SETINFO, &ainfo) < 0) {
+    ai->rate = 0;
+    return 0;
+  }
+  if(ioctl(ai->fn, AUDIO_GETINFO, &ainfo) < 0) {
+    return -1;
+  }
+  ai->rate = ainfo.play.sample_rate;
+  return 0;
+}
+
+static int audio_set_rate(struct audio_info_struct *ai)
+{
+  audio_info_t ainfo;
+
+  if(ai->rate != -1) {
+    AUDIO_INITINFO(&ainfo);
+    ainfo.play.sample_rate = ai->rate;
+    if(ioctl(ai->fn, AUDIO_SETINFO, &ainfo) == -1)
+      return -1;
+    return 0;
+  }
+  return -1;
+}
+
+static int audio_set_channels(struct audio_info_struct *ai)
+{
+  audio_info_t ainfo;
+
+  AUDIO_INITINFO(&ainfo);
+  ainfo.play.channels = ai->channels;
+  if(ioctl(ai->fn, AUDIO_SETINFO, &ainfo) == -1)
+    return -1;
+  return 0;
+}
+
+static void audio_set_format_helper(struct audio_info_struct *ai,audio_info_t *ainfo)
+{
+
+
+  switch(ai->format) {
+    case -1:
+    case AUDIO_FORMAT_SIGNED_16:
+    default:
+      ainfo->play.encoding = AUDIO_ENCODING_LINEAR;
+      ainfo->play.precision = 16;
+      break;
+    case AUDIO_FORMAT_UNSIGNED_8:
+#if defined(SOLARIS) || defined(SPARCLINUX)
+      ainfo->play.encoding = AUDIO_ENCODING_LINEAR8;
+      ainfo->play.precision = 8;
+      break;
+#endif
+#ifdef NETBSD
+      ainfo->play.encoding = AUDIO_ENCODING_LINEAR;
+      ainfo->play.precision = 8;
+      break;
+#endif
+    case AUDIO_FORMAT_SIGNED_8:
+      fprintf(stderr,"Linear signed 8 bit not supported!\n");
+      return;
+    case AUDIO_FORMAT_ULAW_8:
+      ainfo->play.encoding = AUDIO_ENCODING_ULAW;
+      ainfo->play.precision = 8;
+      break;
+    case AUDIO_FORMAT_ALAW_8:
+      ainfo->play.encoding = AUDIO_ENCODING_ALAW;
+      ainfo->play.precision = 8;
+      break;
+  }
+}
+
+
+static int audio_set_format(struct audio_info_struct *ai)
+{
+  audio_info_t ainfo;
+
+  AUDIO_INITINFO(&ainfo);
+  audio_set_format_helper(ai,&ainfo);
+  if(ioctl(ai->fn, AUDIO_SETINFO, &ainfo) == -1)
+    return -1;
+
+  return 0;
+}
 
 int audio_open(struct audio_info_struct *ai)
 {
@@ -100,112 +206,8 @@ int audio_open(struct audio_info_struct *ai)
   return ai->fn;
 }
 
-int audio_reset_parameters(struct audio_info_struct *ai)
-{
-  audio_info_t ainfo;
-
-  AUDIO_INITINFO(&ainfo);
-
-  if(ai->rate != -1)
-    ainfo.play.sample_rate = ai->rate;
-  if(ai->channels >= 0)
-    ainfo.play.channels = ai->channels;
-  audio_set_format_helper(ai,&ainfo);
-
-  if(ioctl(ai->fn, AUDIO_SETINFO, &ainfo) == -1)
-    return -1;
-  return 0;
-}
-
-int audio_rate_best_match(struct audio_info_struct *ai)
-{
-  audio_info_t ainfo;
-  AUDIO_INITINFO(&ainfo);
- 
-  ainfo.play.sample_rate = ai->rate;
-  if(ioctl(ai->fn, AUDIO_SETINFO, &ainfo) < 0) {
-    ai->rate = 0;
-    return 0;
-  }
-  if(ioctl(ai->fn, AUDIO_GETINFO, &ainfo) < 0) {
-    return -1;
-  }
-  ai->rate = ainfo.play.sample_rate;
-  return 0;
-}
-
-int audio_set_rate(struct audio_info_struct *ai)
-{
-  audio_info_t ainfo;
-
-  if(ai->rate != -1) {
-    AUDIO_INITINFO(&ainfo);
-    ainfo.play.sample_rate = ai->rate;
-    if(ioctl(ai->fn, AUDIO_SETINFO, &ainfo) == -1)
-      return -1;
-    return 0;
-  }
-  return -1;
-}
-
-int audio_set_channels(struct audio_info_struct *ai)
-{
-  audio_info_t ainfo;
-
-  AUDIO_INITINFO(&ainfo);
-  ainfo.play.channels = ai->channels;
-  if(ioctl(ai->fn, AUDIO_SETINFO, &ainfo) == -1)
-    return -1;
-  return 0;
-}
-
-static void audio_set_format_helper(struct audio_info_struct *ai,audio_info_t *ainfo)
-{
 
 
-  switch(ai->format) {
-    case -1:
-    case AUDIO_FORMAT_SIGNED_16:
-    default:
-      ainfo->play.encoding = AUDIO_ENCODING_LINEAR;
-      ainfo->play.precision = 16;
-      break;
-    case AUDIO_FORMAT_UNSIGNED_8:
-#if defined(SOLARIS) || defined(SPARCLINUX)
-      ainfo->play.encoding = AUDIO_ENCODING_LINEAR8;
-      ainfo->play.precision = 8;
-      break;
-#endif
-#ifdef NETBSD
-      ainfo->play.encoding = AUDIO_ENCODING_LINEAR;
-      ainfo->play.precision = 8;
-      break;
-#endif
-    case AUDIO_FORMAT_SIGNED_8:
-      fprintf(stderr,"Linear signed 8 bit not supported!\n");
-      return;
-    case AUDIO_FORMAT_ULAW_8:
-      ainfo->play.encoding = AUDIO_ENCODING_ULAW;
-      ainfo->play.precision = 8;
-      break;
-    case AUDIO_FORMAT_ALAW_8:
-      ainfo->play.encoding = AUDIO_ENCODING_ALAW;
-      ainfo->play.precision = 8;
-      break;
-  }
-}
-
-int audio_set_format(struct audio_info_struct *ai)
-{
-  audio_info_t ainfo;
-
-  AUDIO_INITINFO(&ainfo);
-  audio_set_format_helper(ai,&ainfo);
-  if(ioctl(ai->fn, AUDIO_SETINFO, &ainfo) == -1)
-    return -1;
-
-  return 0;
-}
 
 int audio_get_formats(struct audio_info_struct *ai)
 {
diff --git a/src/audio_win32.c b/src/audio_win32.c
index 1e305923..9db3045a 100644
--- a/src/audio_win32.c
+++ b/src/audio_win32.c
@@ -108,31 +108,6 @@ int audio_open(struct audio_info_struct *ai)
    return 0;
 }
 
-int audio_reset_parameters(struct audio_info_struct *ai)
-{
-  return 0;
-}
-
-int audio_rate_best_match(struct audio_info_struct *ai)
-{
-  return 0;
-}
-
-int audio_set_rate(struct audio_info_struct *ai)
-{
-  return 0;
-}
-
-int audio_set_channels(struct audio_info_struct *ai)
-{
-  return 0;
-}
-
-int audio_set_format(struct audio_info_struct *ai)
-{
-  return 0;
-}
-
 int audio_get_formats(struct audio_info_struct *ai)
 {
   return AUDIO_FORMAT_SIGNED_16;
@@ -224,3 +199,8 @@ int audio_close(struct audio_info_struct *ai)
    nBlocks = 0;
    return(0);
 }
+
+void audio_queueflush(struct audio_info_struct *ai)
+{
+}
+
diff --git a/src/dct64.c b/src/dct64.c
index 88fdde0c..144617e2 100644
--- a/src/dct64.c
+++ b/src/dct64.c
@@ -4,9 +4,8 @@
  *
  * -funroll-loops (for gcc) will remove the loops for better performance
  * using loops in the source-code enhances readabillity
- */
-
-/*
+ *
+ *
  * TODO: write an optimized version for the down-sampling modes
  *       (in these modes the bands 16-31 (2:1) or 8-31 (4:1) are zero 
  */
diff --git a/src/getlopt.h b/src/getlopt.h
index 4d80a97c..47360c53 100644
--- a/src/getlopt.h
+++ b/src/getlopt.h
@@ -41,6 +41,7 @@ for .... no flag) */
  *		else
  *			loptarg = &arg
  *			return ((value != 0) ? value : sname)
+ *
  *	bit 1 = 1 - var is a pointer to a char (or string),
  *			and value is interpreted as char
  *	bit 2 = 1 - var is a pointer to int
diff --git a/src/httpget.c b/src/httpget.c
index 55451c1e..19f30e98 100644
--- a/src/httpget.c
+++ b/src/httpget.c
@@ -21,10 +21,7 @@
 #include <errno.h>
 #include <ctype.h>
 
-#ifdef HAVE_CONFIG_H
 #include "config.h"
-#endif
-
 #include "mpg123.h"
 
 #ifndef INADDR_NONE
diff --git a/src/mpg123.c b/src/mpg123.c
index 862c741f..06905613 100644
--- a/src/mpg123.c
+++ b/src/mpg123.c
@@ -7,9 +7,7 @@
  */
 
 
-#ifdef HAVE_CONFIG_H
 #include "config.h"
-#endif
 
 #include <stdlib.h>
 #include <sys/types.h>
@@ -1221,10 +1219,11 @@ tc_hack:
 
 static void print_title(void)
 {
-    fprintf(stderr,"High Performance MPEG 1.0/2.0 Audio Player for Layer 1, 2 and 3.\n");
+    fprintf(stderr,"High Performance MPEG 1.0/2.0 Audio Player for Layers 1, 2 and 3.\n");
     fprintf(stderr,"Version %s. Initially written and copyright by Michael Hipp.\n", PACKAGE_VERSION);
-    fprintf(stderr,"Uses code from various people. See 'README' for more!\n");
-    fprintf(stderr,"THIS SOFTWARE COMES WITH ABSOLUTELY NO WARRANTY! USE AT YOUR OWN RISK!\n");
+    fprintf(stderr,"Uses code from various people, see 'AUTHORS' for full list.\n");
+	fprintf(stderr,"This software comes with ABSOLUTELY NO WARRANTY. For details, see \n");
+	fprintf(stderr,"the enclosed file COPYING for license information (GPL).\n");
 }
 
 static void usage(char *dummy)  /* print syntax & exit */
diff --git a/src/system.c b/src/system.c
index dd69727d..b1e434f4 100644
--- a/src/system.c
+++ b/src/system.c
@@ -100,7 +100,6 @@ static int system_raw_read_head(int f,unsigned long *head)
 static int system_raw_read_word(int f,int *word)
 {
         unsigned char buf[2];
-	int ret;
 
         if(my_read(f,buf,2) != 2) {
 		perror("read_word");
@@ -488,5 +487,6 @@ void main(void)
 	filept = 0;
 	ret = system_init(&rd1);
 	fprintf(stderr,"ret: %d\n",ret);
+	return ret;
 }
 
diff --git a/src/term.c b/src/term.c
index 3dfe2ea2..22da13eb 100644
--- a/src/term.c
+++ b/src/term.c
@@ -1,7 +1,5 @@
 
-#ifdef HAVE_CONFIG_H
 #include "config.h"
-#endif
 
 #ifdef HAVE_TERMIOS
 
diff --git a/src/term.h b/src/term.h
index ba759e5b..c11b12d6 100644
--- a/src/term.h
+++ b/src/term.h
@@ -1,11 +1,8 @@
 
-#ifdef HAVE_CONFIG_H
 #include "config.h"
-#endif
 
 #ifdef HAVE_TERMIOS
 
-
 #define LOOP_CYCLES	0.500000	/* Loop time in sec */
 
 /* 
diff --git a/src/xfermem.c b/src/xfermem.c
index 96d53d73..e216fa35 100644
--- a/src/xfermem.c
+++ b/src/xfermem.c
@@ -7,6 +7,10 @@
  *   See xfermem.h for documentation/description.
  */
 
+#ifdef HAVE_CONFIG
+#include "config.h"
+#endif
+
 #ifndef NOXFERMEM
 
 #include <stdio.h>
@@ -27,12 +31,12 @@
 
 #include "mpg123.h"
 
-#ifndef USE_MMAP
+#ifndef HAVE_MMAP
 #include <sys/ipc.h>
 #include <sys/shm.h>
 #endif
 
-#if defined (USE_MMAP) && defined(MAP_ANONYMOUS) && !defined(MAP_ANON)
+#if defined (HAVE_MMAP) && defined(MAP_ANONYMOUS) && !defined(MAP_ANON)
 #define MAP_ANON MAP_ANONYMOUS
 #endif
 
@@ -40,7 +44,7 @@ void xfermem_init (txfermem **xf, int bufsize, int msize, int skipbuf)
 {
 	int regsize = bufsize + msize + skipbuf + sizeof(txfermem);
 
-#ifdef USE_MMAP
+#ifdef HAVE_MMAP
 #  ifdef MAP_ANON
 	if ((*xf = (txfermem *) mmap(0, regsize, PROT_READ | PROT_WRITE,
 			MAP_ANON | MAP_SHARED, -1, 0)) == (txfermem *) -1) {
@@ -95,7 +99,7 @@ void xfermem_done (txfermem *xf)
 {
 	if(!xf)
 		return;
-#ifdef USE_MMAP
+#ifdef HAVE_MMAP
 	munmap ((caddr_t) xf, xf->size + xf->metasize + sizeof(txfermem));
 #else
 	if (shmdt((void *) xf) == -1) {