removing AFMT_ dependancy

git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@14246 b3059339-0415-0410-9bf9-f77b7e298cf2

30 files changed, 283 insertions, 360 deletions

diff --git a/libao2/Makefile b/libao2/Makefile
index bc3002b1fb..cb1bd674d0 100644
--- a/libao2/Makefile
+++ b/libao2/Makefile
@@ -2,7 +2,7 @@ include config.mak
 
 LIBNAME = libao2.a
 
-SRCS=afmt.c audio_out.c ao_mpegpes.c ao_null.c ao_pcm.c ao_plugin.c pl_delay.c pl_format.c pl_surround.c remez.c pl_resample.c pl_volume.c pl_extrastereo.c pl_volnorm.c pl_eq.c $(OPTIONAL_SRCS)
+SRCS=audio_out.c ao_mpegpes.c ao_null.c ao_pcm.c ao_plugin.c pl_delay.c pl_format.c pl_surround.c remez.c pl_resample.c pl_volume.c pl_extrastereo.c pl_volnorm.c pl_eq.c $(OPTIONAL_SRCS)
 
 OBJS=$(SRCS:.c=.o)
 
diff --git a/libao2/afmt.c b/libao2/afmt.c
deleted file mode 100644
index 95cd817ca2..0000000000
--- a/libao2/afmt.c
+++ /dev/null
@@ -1,90 +0,0 @@
-#include <stdio.h>
-#include <stdlib.h>
-
-#include "config.h"
-#include "afmt.h"
-
-char *audio_out_format_name(int format)
-{
-    switch (format)
-    {
-	case AFMT_MU_LAW:
-	    return("Mu-Law");
-	case AFMT_A_LAW:
-	    return("A-Law");
-	case AFMT_IMA_ADPCM:
-	    return("Ima-ADPCM");
-	case AFMT_S8:
-	    return("Signed 8-bit");
-	case AFMT_U8:
-	    return("Unsigned 8-bit");
-	case AFMT_U16_LE:
-	    return("Unsigned 16-bit (Little-Endian)");
-	case AFMT_U16_BE: 
-	    return("Unsigned 16-bit (Big-Endian)");
-	case AFMT_S16_LE:
-	    return("Signed 16-bit (Little-Endian)");
-	case AFMT_S16_BE:
-	    return("Signed 16-bit (Big-Endian)");
-	case AFMT_MPEG:
-	    return("MPEG (2) audio");
-	case AFMT_AC3:
-	    return("AC3");
-	case AFMT_U32_LE:
-	    return("Unsigned 32-bit (Little-Endian)");
-	case AFMT_U32_BE:
-	    return("Unsigned 32-bit (Big-Endian)");
-	case AFMT_S32_LE:
-	    return("Signed 32-bit (Little-Endian)");
-	case AFMT_S32_BE:
-	    return("Signed 32-bit (Big-Endian)");
-	case AFMT_U24_LE:
-	    return("Unsigned 24-bit (Little-Endian)");
-	case AFMT_U24_BE:
-	    return("Unsigned 24-bit (Big-Endian)");
-	case AFMT_S24_LE:
-	    return("Signed 24-bit (Little-Endian)");
-	case AFMT_S24_BE:
-	    return("Signed 24-bit (Big-Endian)");
-	case AFMT_FLOAT:
-	    return("Floating Point");
-    }
-    return("Unknown");
-}
-
-// return number of bits for 1 sample in one channel, or 8 bits for compressed
-int audio_out_format_bits(int format){
-    switch (format)
-    {
-	case AFMT_S16_LE:
-	case AFMT_S16_BE:
-	case AFMT_U16_LE:
-	case AFMT_U16_BE: 
-	return 16;//16 bits
-
-	case AFMT_S32_LE:
-	case AFMT_S32_BE:
-	case AFMT_U32_LE:
-	case AFMT_U32_BE:
-	case AFMT_FLOAT:
-	return 32;
-
-	case AFMT_S24_LE:
-	case AFMT_S24_BE:
-	case AFMT_U24_LE:
-	case AFMT_U24_BE:
-	return 24;
-	
-	case AFMT_MU_LAW:
-	case AFMT_A_LAW:
-	case AFMT_IMA_ADPCM:
-	case AFMT_S8:
-	case AFMT_U8:
-	case AFMT_MPEG:
-	case AFMT_AC3:
-	default:
-	    return 8;//default 1 byte
-	
-    }
-    return 8;
-}
diff --git a/libao2/afmt.h b/libao2/afmt.h
deleted file mode 100644
index 04dc762c1b..0000000000
--- a/libao2/afmt.h
+++ /dev/null
@@ -1,84 +0,0 @@
-
-/* Defines that AFMT_ stuff */
-
-#ifdef	HAVE_SYS_SOUNDCARD_H
-#include <sys/soundcard.h>	/* For AFMT_* on linux */
-#else 
-#ifdef HAVE_SOUNDCARD_H
-#include <soundcard.h> /* OpenBSD have this instead of <sys/soundcard> */
-#endif
-#endif
-
-#include "config.h" /* for native endianness */
-
-/* standard, old OSS audio formats */
-#ifndef AFMT_MU_LAW
-#	define AFMT_MU_LAW		0x00000001
-#	define AFMT_A_LAW		0x00000002
-#	define AFMT_IMA_ADPCM		0x00000004
-#	define AFMT_U8			0x00000008
-#	define AFMT_S16_LE		0x00000010	/* Little endian signed 16*/
-#	define AFMT_S16_BE		0x00000020	/* Big endian signed 16 */
-#	define AFMT_S8			0x00000040
-#	define AFMT_U16_LE		0x00000080	/* Little endian U16 */
-#	define AFMT_U16_BE		0x00000100	/* Big endian U16 */
-#endif
-
-#ifndef AFMT_MPEG
-#	define AFMT_MPEG		0x00000200	/* MPEG (2) audio */
-#endif
-
-#ifndef AFMT_AC3
-#	define AFMT_AC3			0x00000400	/* Dolby Digital AC3 */
-#endif
-
-/* 24 bit formats from the linux kernel */
-#ifndef AFMT_S24_LE
-
-// FreeBSD fix...
-#if AFMT_S32_LE == 0x1000
-
-#define AFMT_S24_LE		0x00010000
-#define AFMT_S24_BE		0x00020000
-#define AFMT_U24_LE		0x00040000
-#define AFMT_U24_BE		0x00080000
-
-#else
-
-#define AFMT_S24_LE		0x00000800
-#define AFMT_S24_BE		0x00001000
-#define AFMT_U24_LE		0x00002000
-#define AFMT_U24_BE		0x00004000
-
-#endif
-
-#endif
-
-/* 32 bit formats from the linux kernel */
-#ifndef AFMT_S32_LE
-#define AFMT_S32_LE		0x00008000
-#define AFMT_S32_BE		0x00010000
-#define AFMT_U32_LE		0x00020000
-#define AFMT_U32_BE		0x00040000
-#endif
-
-/* native endian formats */
-#ifndef	AFMT_S16_NE
-# if WORDS_BIGENDIAN
-#  define AFMT_S16_NE	AFMT_S16_BE
-#  define AFMT_S24_NE	AFMT_S24_BE
-#  define AFMT_S32_NE	AFMT_S32_BE
-# else
-#  define AFMT_S16_NE	AFMT_S16_LE
-#  define AFMT_S24_NE	AFMT_S24_LE
-#  define AFMT_S32_NE	AFMT_S32_LE
-# endif
-#endif
-
-#ifndef AFMT_FLOAT
-# define AFMT_FLOAT               0x00100000
-#endif
-
-/* for formats that don't have a corresponding AFMT_* type,
- * use the flags from libaf/af_format.h or'ed with this */
-#define AFMT_AF_FLAGS             0x70000000
diff --git a/libao2/ao_alsa.c b/libao2/ao_alsa.c
index 2aa6cdaebe..539fd0fa7c 100644
--- a/libao2/ao_alsa.c
+++ b/libao2/ao_alsa.c
@@ -37,7 +37,7 @@
 
 #include "audio_out.h"
 #include "audio_out_internal.h"
-#include "afmt.h"
+#include "libaf/af_format.h"
 
 static ao_info_t info = 
 {
@@ -125,7 +125,7 @@ static int control(int cmd, void *arg)
       }
       if(mixer_device) card = mixer_device;
 
-      if(ao_data.format == AFMT_AC3)
+      if(ao_data.format == AF_FORMAT_AC3)
 	return CONTROL_TRUE;
 
       //allocate simple id
@@ -275,42 +275,42 @@ static int init(int rate_hz, int channels, int format, int flags)
 
     switch (format)
       {
-      case AFMT_S8:
+      case AF_FORMAT_S8:
 	alsa_format = SND_PCM_FORMAT_S8;
 	break;
-      case AFMT_U8:
+      case AF_FORMAT_U8:
 	alsa_format = SND_PCM_FORMAT_U8;
 	break;
-      case AFMT_U16_LE:
+      case AF_FORMAT_U16_LE:
 	alsa_format = SND_PCM_FORMAT_U16_LE;
 	break;
-      case AFMT_U16_BE:
+      case AF_FORMAT_U16_BE:
 	alsa_format = SND_PCM_FORMAT_U16_BE;
 	break;
 #ifndef WORDS_BIGENDIAN
-      case AFMT_AC3:
+      case AF_FORMAT_AC3:
 #endif
-      case AFMT_S16_LE:
+      case AF_FORMAT_S16_LE:
 	alsa_format = SND_PCM_FORMAT_S16_LE;
 	break;
 #ifdef WORDS_BIGENDIAN
-      case AFMT_AC3:
+      case AF_FORMAT_AC3:
 #endif
-      case AFMT_S16_BE:
+      case AF_FORMAT_S16_BE:
 	alsa_format = SND_PCM_FORMAT_S16_BE;
 	break;
-      case AFMT_S32_LE:
+      case AF_FORMAT_S32_LE:
 	alsa_format = SND_PCM_FORMAT_S32_LE;
 	break;
-      case AFMT_S32_BE:
+      case AF_FORMAT_S32_BE:
 	alsa_format = SND_PCM_FORMAT_S32_BE;
 	break;
-      case AFMT_FLOAT:
+      case AF_FORMAT_FLOAT_LE:
 	alsa_format = SND_PCM_FORMAT_FLOAT_LE;
 	break;
 
       default:
-	alsa_format = SND_PCM_FORMAT_MPEG; //? default should be -1
+	alsa_format = SND_PCM_FORMAT_MPEG2; //? default should be -1
 	break;
       }
     
@@ -412,7 +412,7 @@ static int init(int rate_hz, int channels, int format, int flags)
      * while opening the abstract alias for the spdif subdevice
      * 'iec958'
      */
-    if (format == AFMT_AC3) {
+    if (format == AF_FORMAT_AC3) {
       unsigned char s[4];
 
       switch (channels) {
@@ -590,7 +590,7 @@ static int init(int rate_hz, int channels, int format, int flags)
 		"alsa-init: format %s are not supported by hardware, trying default\n", 
 		audio_out_format_name(format));
          alsa_format = SND_PCM_FORMAT_S16_LE;
-         ao_data.format = AFMT_S16_LE;
+         ao_data.format = AF_FORMAT_S16_LE;
          ao_data.bps = channels * rate_hz * 2;
       }
 
diff --git a/libao2/ao_alsa5.c b/libao2/ao_alsa5.c
index 68f2cbcaa5..ed073f55ae 100644
--- a/libao2/ao_alsa5.c
+++ b/libao2/ao_alsa5.c
@@ -13,7 +13,7 @@
 
 #include "audio_out.h"
 #include "audio_out_internal.h"
-#include "afmt.h"
+#include "libaf/af_format.h"
 
 #include "mp_msg.h"
 #include "help_mp.h"
@@ -75,28 +75,28 @@ static int init(int rate_hz, int channels, int format, int flags)
     memset(&alsa_format, 0, sizeof(alsa_format));
     switch (format)
     {
-	case AFMT_S8:
+	case AF_FORMAT_S8:
 	    alsa_format.format = SND_PCM_SFMT_S8;
 	    break;
-	case AFMT_U8:
+	case AF_FORMAT_U8:
 	    alsa_format.format = SND_PCM_SFMT_U8;
 	    break;
-	case AFMT_U16_LE:
+	case AF_FORMAT_U16_LE:
 	    alsa_format.format = SND_PCM_SFMT_U16_LE;
 	    break;
-	case AFMT_U16_BE:
+	case AF_FORMAT_U16_BE:
 	    alsa_format.format = SND_PCM_SFMT_U16_BE;
 	    break;
 #ifndef WORDS_BIGENDIAN
-	case AFMT_AC3:
+	case AF_FORMAT_AC3:
 #endif
-	case AFMT_S16_LE:
+	case AF_FORMAT_S16_LE:
 	    alsa_format.format = SND_PCM_SFMT_S16_LE;
 	    break;
 #ifdef WORDS_BIGENDIAN
-	case AFMT_AC3:
+	case AF_FORMAT_AC3:
 #endif
-	case AFMT_S16_BE:
+	case AF_FORMAT_S16_BE:
 	    alsa_format.format = SND_PCM_SFMT_S16_BE;
 	    break;
 	default:
diff --git a/libao2/ao_arts.c b/libao2/ao_arts.c
index 8d2918d74f..ad19b9845b 100644
--- a/libao2/ao_arts.c
+++ b/libao2/ao_arts.c
@@ -12,12 +12,12 @@
 
 #include "audio_out.h"
 #include "audio_out_internal.h"
-#include "afmt.h"
+#include "libaf/af_format.h"
 #include "config.h"
 #include "mp_msg.h"
 #include "help_mp.h"
 
-#define OBTAIN_BITRATE(a) (((a != AFMT_U8) && (a != AFMT_S8)) ? 16 : 8)
+#define OBTAIN_BITRATE(a) (((a != AF_FORMAT_U8) && (a != AF_FORMAT_S8)) ? 16 : 8)
 
 /* Feel free to experiment with the following values: */
 #define ARTS_PACKETS 10 /* Number of audio packets */
@@ -60,12 +60,12 @@ static int init(int rate_hz, int channels, int format, int flags)
 	 * using mplayer's audio filters.
 	 */
 	switch (format) {
-	case AFMT_U8:
-	case AFMT_S8:
-	    format = AFMT_U8;
+	case AF_FORMAT_U8:
+	case AF_FORMAT_S8:
+	    format = AF_FORMAT_U8;
 	    break;
 	default:
-	    format = AFMT_S16_LE;    /* artsd always expects little endian?*/
+	    format = AF_FORMAT_S16_LE;    /* artsd always expects little endian?*/
 	    break;
 	}
 
@@ -74,7 +74,7 @@ static int init(int rate_hz, int channels, int format, int flags)
 	ao_data.samplerate = rate_hz;
 	ao_data.bps = (rate_hz*channels);
 
-	if(format != AFMT_U8 && format != AFMT_S8)
+	if(format != AF_FORMAT_U8 && format != AF_FORMAT_S8)
 		ao_data.bps*=2;
 
 	stream=arts_play_stream(rate_hz, OBTAIN_BITRATE(format), channels, "MPlayer");
diff --git a/libao2/ao_dsound.c b/libao2/ao_dsound.c
index 0344c7410e..39ab901df8 100644
--- a/libao2/ao_dsound.c
+++ b/libao2/ao_dsound.c
@@ -28,7 +28,7 @@
 #define DIRECTSOUND_VERSION 0x0600
 #include <dsound.h>
 
-#include "afmt.h"
+#include "libaf/af_format.h"
 #include "audio_out.h"
 #include "audio_out_internal.h"
 #include "mp_msg.h"
@@ -265,7 +265,7 @@ static int write_buffer(unsigned char *data, int len)
   
   if (SUCCEEDED(res)) 
   {
-  	if( (ao_data.channels == 6) && (ao_data.format!=AFMT_AC3) ) {
+  	if( (ao_data.channels == 6) && (ao_data.format!=AF_FORMAT_AC3) ) {
   	    // reorder channels while writing to pointers.
   	    // it's this easy because buffer size and len are always
   	    // aligned to multiples of channels*bytespersample
@@ -366,14 +366,14 @@ static int init(int rate, int channels, int format, int flags)
 
 	//check if the format is supported in general
 	switch(format){
-		case AFMT_AC3:
-		case AFMT_S24_LE:
-		case AFMT_S16_LE:
-		case AFMT_S8:
+		case AF_FORMAT_AC3:
+		case AF_FORMAT_S24_LE:
+		case AF_FORMAT_S16_LE:
+		case AF_FORMAT_S8:
 			break;
 		default:
 			mp_msg(MSGT_AO, MSGL_V,"ao_dsound: format %s not supported defaulting to Signed 16-bit Little-Endian\n",audio_out_format_name(format));
-			format=AFMT_S16_LE;
+			format=AF_FORMAT_S16_LE;
 	}   	
 	//fill global ao_data
 	ao_data.channels = channels;
@@ -389,7 +389,7 @@ static int init(int rate, int channels, int format, int flags)
 	wformat.Format.cbSize          = (channels > 2) ? sizeof(WAVEFORMATEXTENSIBLE)-sizeof(WAVEFORMATEX) : 0;
 	wformat.Format.nChannels       = channels;
 	wformat.Format.nSamplesPerSec  = rate;
-	if (format == AFMT_AC3) {
+	if (format == AF_FORMAT_AC3) {
 		wformat.Format.wFormatTag      = WAVE_FORMAT_DOLBY_AC3_SPDIF;
 		wformat.Format.wBitsPerSample  = 16;
 		wformat.Format.nBlockAlign     = 4;
diff --git a/libao2/ao_dxr2.c b/libao2/ao_dxr2.c
index 7ec4d2c499..f495d12ed4 100644
--- a/libao2/ao_dxr2.c
+++ b/libao2/ao_dxr2.c
@@ -11,8 +11,7 @@
 
 #include "audio_out.h"
 #include "audio_out_internal.h"
-
-#include "afmt.h"
+#include "libaf/af_format.h"
 
 
 static ao_info_t info =
@@ -158,9 +157,9 @@ static int get_space(){
 // return: number of bytes played
 static int play(void* data,int len,int flags){
   // MPEG and AC3 don't work :-(
-    if(ao_data.format==AFMT_MPEG)
+    if(ao_data.format==AF_FORMAT_MPEG2)
 	dxr2_send_packet(data,len,0xC0,ao_data.pts);
-    else if(ao_data.format==AFMT_AC3)
+    else if(ao_data.format==AF_FORMAT_AC3)
       	dxr2_send_packet(data,len,0x80,ao_data.pts);
     else {
 	int i;
diff --git a/libao2/ao_esd.c b/libao2/ao_esd.c
index 11001062d6..a850ad6009 100644
--- a/libao2/ao_esd.c
+++ b/libao2/ao_esd.c
@@ -32,7 +32,7 @@
 
 #include "audio_out.h"
 #include "audio_out_internal.h"
-#include "afmt.h"
+#include "libaf/af_format.h"
 #include "config.h"
 #include "mp_msg.h"
 #include "help_mp.h"
@@ -200,14 +200,14 @@ static int init(int rate_hz, int channels, int format, int flags)
 
     /* EsounD can play 8bit unsigned and 16bit signed native */
     switch (format) {
-    case AFMT_S8:
-    case AFMT_U8:
+    case AF_FORMAT_S8:
+    case AF_FORMAT_U8:
 	esd_fmt |= ESD_BITS8;
-	ao_data.format = AFMT_U8;
+	ao_data.format = AF_FORMAT_U8;
 	break;
     default:
 	esd_fmt |= ESD_BITS16;
-	ao_data.format = AFMT_S16_NE;
+	ao_data.format = AF_FORMAT_S16_NE;
 	bytes_per_sample *= 2;
 	break;
     }
diff --git a/libao2/ao_jack.c b/libao2/ao_jack.c
index 0f4ce43061..6e11c9752d 100644
--- a/libao2/ao_jack.c
+++ b/libao2/ao_jack.c
@@ -14,7 +14,7 @@
 
 #include "audio_out.h"
 #include "audio_out_internal.h"
-#include "afmt.h"
+#include "libaf/af_format.h"
 #include "config.h"
 #include "mp_msg.h"
 
@@ -146,14 +146,14 @@ static int init(int rate_hz, int channels, int format, int flags)
 	}
 
 	switch (format) {
-		case AFMT_U8:
-		case AFMT_S8:
-			format = AFMT_U8;
+		case AF_FORMAT_U8:
+		case AF_FORMAT_S8:
+			format = AF_FORMAT_U8;
 			bits_per_sample = 8;
 			m = 1;
 			break;
 		default:
-			format = AFMT_S16_LE;
+			format = AF_FORMAT_S16_LE;
 			bits_per_sample = 16;
 			m = 2;
 			break;
diff --git a/libao2/ao_macosx.c b/libao2/ao_macosx.c
index b13fcac4c9..8ab0ab2b3c 100644
--- a/libao2/ao_macosx.c
+++ b/libao2/ao_macosx.c
@@ -47,7 +47,7 @@
 
 #include "audio_out.h"
 #include "audio_out_internal.h"
-#include "afmt.h"
+#include "libaf/af_format.h"
 
 static ao_info_t info =
   {
@@ -252,7 +252,7 @@ static int init(int rate,int channels,int format,int flags)
     if (ao->outputStreamBasicDescription.mFormatID == kAudioFormatLinearPCM) {
       uint32_t flags = ao->outputStreamBasicDescription.mFormatFlags;
       if (flags & kAudioFormatFlagIsFloat) {
-	ao_data.format = AFMT_FLOAT;
+	ao_data.format = AF_FORMAT_FLOAT_NE;
       } else {
 	ao_msg(MSGT_AO,MSGL_WARN, "Unsupported audio output "
 	       "format %d. Please report this to the developer\n",
diff --git a/libao2/ao_mpegpes.c b/libao2/ao_mpegpes.c
index be9ad4a7f2..925d78b609 100644
--- a/libao2/ao_mpegpes.c
+++ b/libao2/ao_mpegpes.c
@@ -15,7 +15,7 @@
 #include "audio_out.h"
 #include "audio_out_internal.h"
 
-#include "afmt.h"
+#include "libaf/af_format.h"
 
 #include "mp_msg.h"
 #include "help_mp.h"
@@ -96,14 +96,14 @@ static int init(int rate,int channels,int format,int flags){
     ao_data.channels=2;
     ao_data.outburst=2000;
     switch(format){
-	case AFMT_S16_LE:
-	case AFMT_S16_BE:
-	case AFMT_MPEG:
-	case AFMT_AC3:
+	case AF_FORMAT_S16_LE:
+	case AF_FORMAT_S16_BE:
+	case AF_FORMAT_MPEG2:
+	case AF_FORMAT_AC3:
 	    ao_data.format=format;
 	    break;
 	default:
-	    ao_data.format=AFMT_S16_BE;
+	    ao_data.format=AF_FORMAT_S16_BE;
     }
     
 retry:
@@ -174,14 +174,14 @@ static int get_space(){
 // return: number of bytes played
 static int play(void* data,int len,int flags){
 //    printf("\nao_mpegpes: play(%d) freq=%d\n",len,freq_id);
-    if(ao_data.format==AFMT_MPEG)
+    if(ao_data.format==AF_FORMAT_MPEG2)
 	send_pes_packet(data,len,0x1C0,ao_data.pts);
     else {
 	int i;
 	unsigned short *s=data;
 //	if(len>2000) len=2000;
 //	printf("ao_mpegpes: len=%d  \n",len);
-	if(ao_data.format==AFMT_S16_LE || ao_data.format==AFMT_AC3)
+	if(ao_data.format==AF_FORMAT_S16_LE || ao_data.format==AF_FORMAT_AC3)
 	    for(i=0;i<len/2;i++) s[i]=(s[i]>>8)|(s[i]<<8); // le<->be
 	send_lpcm_packet(data,len,0xA0,ao_data.pts,freq_id);
     }
diff --git a/libao2/ao_nas.c b/libao2/ao_nas.c
index 6f77bcbd9e..5955ac1b89 100644
--- a/libao2/ao_nas.c
+++ b/libao2/ao_nas.c
@@ -37,7 +37,7 @@
 
 #include "audio_out.h"
 #include "audio_out_internal.h"
-#include "afmt.h"
+#include "libaf/af_format.h"
 
 #define NAS_FRAG_SIZE 4096
 
@@ -312,22 +312,22 @@ static AuDeviceID nas_find_device(AuServer *aud, int nch)
 static unsigned int nas_aformat_to_auformat(unsigned int *format)
 {
 	switch (*format) {
-	case	AFMT_U8:
+	case	AF_FORMAT_U8:
 		return AuFormatLinearUnsigned8;
-	case	AFMT_S8:
+	case	AF_FORMAT_S8:
 		return AuFormatLinearSigned8;
-	case	AFMT_U16_LE:
+	case	AF_FORMAT_U16_LE:
 		return AuFormatLinearUnsigned16LSB;
-	case	AFMT_U16_BE:
+	case	AF_FORMAT_U16_BE:
 		return AuFormatLinearUnsigned16MSB;
-	case	AFMT_S16_LE:
+	case	AF_FORMAT_S16_LE:
 		return AuFormatLinearSigned16LSB;
-	case	AFMT_S16_BE:
+	case	AF_FORMAT_S16_BE:
 		return AuFormatLinearSigned16MSB;
-	case	AFMT_MU_LAW:
+	case	AF_FORMAT_MU_LAW:
 		return AuFormatULAW8;
 	default:
-		*format=AFMT_S16_NE;
+		*format=AF_FORMAT_S16_NE;
 		return nas_aformat_to_auformat(format);
 	}
 }
diff --git a/libao2/ao_null.c b/libao2/ao_null.c
index 59d115a400..80f3f6c351 100644
--- a/libao2/ao_null.c
+++ b/libao2/ao_null.c
@@ -2,7 +2,7 @@
 #include <stdlib.h>
 #include <sys/time.h>
 
-#include "afmt.h"
+#include "libaf/af_format.h"
 #include "audio_out.h"
 #include "audio_out_internal.h"
 
@@ -55,7 +55,7 @@ static int init(int rate,int channels,int format,int flags){
     ao_data.samplerate=rate;
     ao_data.format=format;
     ao_data.bps=channels*rate;
-    if (format != AFMT_U8 && format != AFMT_S8)
+    if (format != AF_FORMAT_U8 && format != AF_FORMAT_S8)
 	ao_data.bps*=2; 
     buffer=0;
     gettimeofday(&last_tv, 0);
diff --git a/libao2/ao_oss.c b/libao2/ao_oss.c
index 8028835b81..7fa33e4603 100644
--- a/libao2/ao_oss.c
+++ b/libao2/ao_oss.c
@@ -9,14 +9,21 @@
 #include <fcntl.h>
 #include <errno.h>
 #include <string.h>
-//#include <sys/soundcard.h>
 
 #include "config.h"
 #include "mp_msg.h"
 #include "mixer.h"
 #include "help_mp.h"
 
-#include "afmt.h"
+#ifdef HAVE_SYS_SOUNDCARD_H
+#include <sys/soundcard.h>
+#else
+#ifdef HAVE_SOUNDCARD_H
+#include <soundcard.h>
+#endif
+#endif
+
+#include "../libaf/af_format.h"
 
 #include "audio_out.h"
 #include "audio_out_internal.h"
@@ -33,6 +40,86 @@ static ao_info_t info =
 
 LIBAO_EXTERN(oss)
 
+static int format2oss(int format)
+{
+    switch(format)
+    {
+    case AF_FORMAT_U8: return AFMT_U8;
+    case AF_FORMAT_S8: return AFMT_S8;
+    case AF_FORMAT_U16_LE: return AFMT_U16_LE;
+    case AF_FORMAT_U16_BE: return AFMT_U16_BE;
+    case AF_FORMAT_S16_LE: return AFMT_S16_LE;
+    case AF_FORMAT_S16_BE: return AFMT_S16_BE;
+#ifdef AFMT_S24_LE
+    case AF_FORMAT_U24_LE: return AFMT_U24_LE;
+    case AF_FORMAT_U24_BE: return AFMT_U24_BE;
+    case AF_FORMAT_S24_LE: return AFMT_S24_LE;
+    case AF_FORMAT_S24_BE: return AFMT_S24_BE;
+#endif
+#ifdef AFMT_S32_LE
+    case AF_FORMAT_U32_LE: return AFMT_U32_LE;
+    case AF_FORMAT_U32_BE: return AFMT_U32_BE;
+    case AF_FORMAT_S32_LE: return AFMT_S32_LE;
+    case AF_FORMAT_S32_BE: return AFMT_S32_BE;
+#endif
+#ifdef AFMT_FLOAT
+    case AF_FORMAT_FLOAT_NE: return AFMT_FLOAT;
+#endif
+    // SPECIALS
+    case AF_FORMAT_MU_LAW: return AFMT_MU_LAW;
+    case AF_FORMAT_A_LAW: return AFMT_A_LAW;
+    case AF_FORMAT_IMA_ADPCM: return AFMT_IMA_ADPCM;
+#ifdef AFMT_MPEG
+    case AF_FORMAT_MPEG2: return AFMT_MPEG;
+#endif
+#ifdef AFMT_AC3
+    case AF_FORMAT_AC3: return AFMT_AC3;
+#endif
+    }
+    printf("Unknown format: %x\n", format);
+    return -1;
+}
+
+static int oss2format(int format)
+{
+    switch(format)
+    {
+    case AFMT_U8: return AF_FORMAT_U8;
+    case AFMT_S8: return AF_FORMAT_S8;
+    case AFMT_U16_LE: return AF_FORMAT_U16_LE;
+    case AFMT_U16_BE: return AF_FORMAT_U16_BE;
+    case AFMT_S16_LE: return AF_FORMAT_S16_LE;
+    case AFMT_S16_BE: return AF_FORMAT_S16_BE;
+#ifdef AFMT_S24_LE
+    case AFMT_U24_LE: return AF_FORMAT_U24_LE;
+    case AFMT_U24_BE: return AF_FORMAT_U24_BE;
+    case AFMT_S24_LE: return AF_FORMAT_S24_LE;
+    case AFMT_S24_BE: return AF_FORMAT_S24_BE;
+#endif
+#ifdef AFMT_S32_LE
+    case AFMT_U32_LE: return AF_FORMAT_U32_LE;
+    case AFMT_U32_BE: return AF_FORMAT_U32_BE;
+    case AFMT_S32_LE: return AF_FORMAT_S32_LE;
+    case AFMT_S32_BE: return AF_FORMAT_S32_BE;
+#endif
+#ifdef AFMT_FLOAT
+    case AFMT_FLOAT: return AF_FORMAT_FLOAT_NE;
+#endif
+    // SPECIALS
+    case AFMT_MU_LAW: return AF_FORMAT_MU_LAW;
+    case AFMT_A_LAW: return AF_FORMAT_A_LAW;
+    case AFMT_IMA_ADPCM: return AF_FORMAT_IMA_ADPCM;
+#ifdef AFMT_MPEG
+    case AFMT_MPEG: return AF_FORMAT_MPEG2;
+#endif
+#ifdef AFMT_AC3
+    case AFMT_AC3: return AF_FORMAT_AC3;
+#endif
+    }
+    printf("Unknown format: %x\n", format);
+    return -1;
+}
+
 static char *dsp=PATH_DEV_DSP;
 static audio_buf_info zz;
 static int audio_fd=-1;
@@ -57,7 +144,7 @@ static int control(int cmd,void *arg){
 	    ao_control_vol_t *vol = (ao_control_vol_t *)arg;
 	    int fd, v, devs;
 
-	    if(ao_data.format == AFMT_AC3)
+	    if(ao_data.format == AF_FORMAT_AC3)
 		return CONTROL_TRUE;
     
 	    if ((fd = open(oss_mixer_device, O_RDONLY)) > 0)
@@ -95,9 +182,10 @@ static int control(int cmd,void *arg){
 // return: 1=success 0=fail
 static int init(int rate,int channels,int format,int flags){
   char *mixer_channels [SOUND_MIXER_NRDEVICES] = SOUND_DEVICE_NAMES;
+  int oss_format;
 
-  mp_msg(MSGT_AO,MSGL_V,"ao2: %d Hz  %d chans  %s\n",rate,channels,
-    audio_out_format_name(format));
+//  mp_msg(MSGT_AO,MSGL_V,"ao2: %d Hz  %d chans  %s\n",rate,channels,
+//    audio_out_format_name(format));
 
   if (ao_subdevice)
     dsp = ao_subdevice;
@@ -160,32 +248,39 @@ static int init(int rate,int channels,int format,int flags){
   fcntl(audio_fd, F_SETFD, FD_CLOEXEC);
 #endif
   
-  if(format == AFMT_AC3) {
+  if(format == AF_FORMAT_AC3) {
     ao_data.samplerate=rate;
     ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
   }
 
 ac3_retry:  
   ao_data.format=format;
-  if( ioctl(audio_fd, SNDCTL_DSP_SETFMT, &ao_data.format)<0 ||
-      ao_data.format != format) if(format == AFMT_AC3){
+  oss_format=format2oss(format);
+  if (oss_format == -1) return 0;
+  if( ioctl(audio_fd, SNDCTL_DSP_SETFMT, &oss_format)<0 ||
+      oss_format != format2oss(format)) if(format == AF_FORMAT_AC3){
     mp_msg(MSGT_AO,MSGL_WARN, MSGTR_AO_OSS_CantSetAC3, dsp);
 #ifdef WORDS_BIGENDIAN
-    format=AFMT_S16_BE;
+    oss_format=AFMT_S16_BE;
+    format=AF_FORMAT_S16_BE;
 #else
-    format=AFMT_S16_LE;
+    oss_format=AFMT_S16_LE;
+    format=AF_FORMAT_S16_LE;
 #endif
     goto ac3_retry;
   }
-  mp_msg(MSGT_AO,MSGL_V,"audio_setup: sample format: %s (requested: %s)\n",
-    audio_out_format_name(ao_data.format), audio_out_format_name(format));
+//  mp_msg(MSGT_AO,MSGL_V,"audio_setup: sample format: %s (requested: %s)\n",
+//    audio_out_format_name(ao_data.format), audio_out_format_name(format));
 #if 0
-  if(ao_data.format!=format)
+  if(oss_format!=format2oss(format))
 	mp_msg(MSGT_AO,MSGL_WARN,"WARNING! Your soundcard does NOT support %s sample format! Broken audio or bad playback speed are possible! Try with '-aop list=format'\n",audio_out_format_name(format));
 #endif
+
+  ao_data.format = oss2format(oss_format);
+  if (ao_data.format == -1) return 0;
   
   ao_data.channels = channels;
-  if(format != AFMT_AC3) {
+  if(format != AF_FORMAT_AC3) {
     // We only use SNDCTL_DSP_CHANNELS for >2 channels, in case some drivers don't have it
     if (ao_data.channels > 2) {
       if ( ioctl(audio_fd, SNDCTL_DSP_CHANNELS, &ao_data.channels) == -1 ||
@@ -253,7 +348,7 @@ ac3_retry:
   }
 
   ao_data.bps=ao_data.channels;
-  if(ao_data.format != AFMT_U8 && ao_data.format != AFMT_S8)
+  if(ao_data.format != AF_FORMAT_U8 && ao_data.format != AF_FORMAT_S8)
     ao_data.bps*=2;
 
   ao_data.outburst-=ao_data.outburst % ao_data.bps; // round down
@@ -280,6 +375,7 @@ static void uninit(int immed){
 
 // stop playing and empty buffers (for seeking/pause)
 static void reset(){
+  int oss_format;
     uninit(1);
     audio_fd=open(dsp, O_WRONLY);
     if(audio_fd < 0){
@@ -291,8 +387,9 @@ static void reset(){
   fcntl(audio_fd, F_SETFD, FD_CLOEXEC);
 #endif
 
-  ioctl (audio_fd, SNDCTL_DSP_SETFMT, &ao_data.format);
-  if(ao_data.format != AFMT_AC3) {
+  oss_format = format2oss(ao_data.format);
+  ioctl (audio_fd, SNDCTL_DSP_SETFMT, &oss_format);
+  if(ao_data.format != AF_FORMAT_AC3) {
     if (ao_data.channels > 2)
       ioctl (audio_fd, SNDCTL_DSP_CHANNELS, &ao_data.channels);
     else {
diff --git a/libao2/ao_pcm.c b/libao2/ao_pcm.c
index d1cf65f2a0..609f79c9f0 100644
--- a/libao2/ao_pcm.c
+++ b/libao2/ao_pcm.c
@@ -5,7 +5,7 @@
 #include <string.h>
 
 #include "bswap.h"
-#include "afmt.h"
+#include "libaf/af_format.h"
 #include "audio_out.h"
 #include "audio_out_internal.h"
 #include "mp_msg.h"
@@ -89,12 +89,12 @@ static int init(int rate,int channels,int format,int flags){
 
 	bits=8;
 	switch(format){
-	case AFMT_S8:
-	    format=AFMT_U8;
-	case AFMT_U8:
+	case AF_FORMAT_S8:
+	    format=AF_FORMAT_U8;
+	case AF_FORMAT_U8:
 	    break;
 	default:
-	    format=AFMT_S16_LE;
+	    format=AF_FORMAT_S16_LE;
 	    bits=16;
 	    break;
 	}
@@ -114,9 +114,9 @@ static int init(int rate,int channels,int format,int flags){
 	wavhdr.data_length=le2me_32(0x7ffff000);
 	wavhdr.file_length = wavhdr.data_length + sizeof(wavhdr) - 8;
 
-	mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_PCM_FileInfo, ao_outputfilename, 
-	       (ao_pcm_waveheader?"WAVE":"RAW PCM"), rate, 
-	       (channels > 1) ? "Stereo" : "Mono", audio_out_format_name(format));
+//	mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_PCM_FileInfo, ao_outputfilename, 
+//	       (ao_pcm_waveheader?"WAVE":"RAW PCM"), rate, 
+//	       (channels > 1) ? "Stereo" : "Mono", audio_out_format_name(format));
 	mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_PCM_HintInfo);
 
 	fp = fopen(ao_outputfilename, "wb");
diff --git a/libao2/ao_plugin.c b/libao2/ao_plugin.c
index 357d338743..0da8959c3e 100644
--- a/libao2/ao_plugin.c
+++ b/libao2/ao_plugin.c
@@ -6,7 +6,7 @@
 
 #include "mp_msg.h"
 #include "help_mp.h"
-#include "afmt.h"
+#include "libaf/af_format.h"
 #include "audio_out.h"
 #include "audio_out_internal.h"
 
@@ -140,7 +140,7 @@ static int init(int rate,int channels,int format,int flags){
 
   ao_plugin_local_data.format=format;
   ao_plugin_local_data.channels=channels;
-  ao_plugin_local_data.bpm=audio_out_format_bits(format);
+  ao_plugin_local_data.bpm=af_fmt2bits(format);
 
   ao_plugin_data.rate=rate;
   ao_plugin_data.channels=channels;
@@ -168,7 +168,7 @@ static int init(int rate,int channels,int format,int flags){
   // Calculate bps
   ao_plugin_local_data.bps=(float)(ao_plugin_data.rate * 
 				   ao_plugin_data.channels);
-  ao_plugin_local_data.bps*=audio_out_format_bits(ao_plugin_data.format)/8;
+  ao_plugin_local_data.bps*=af_fmt2bits(ao_plugin_data.format)/8;
 
   // This should never happen but check anyway 
   if(NULL==ao_plugin_local_data.driver)
diff --git a/libao2/ao_polyp.c b/libao2/ao_polyp.c
index ef072d2381..e75d05aaf4 100644
--- a/libao2/ao_polyp.c
+++ b/libao2/ao_polyp.c
@@ -6,7 +6,7 @@
 
 #include "audio_out.h"
 #include "audio_out_internal.h"
-#include "afmt.h"
+#include "libaf/af_format.h"
 #include "config.h"
 #include "mp_msg.h"
 
@@ -91,16 +91,16 @@ static int init(int rate_hz, int channels, int format, int flags) {
     ss.rate = rate_hz;
 
     switch (format) {
-        case AFMT_U8:
+        case AF_FORMAT_U8:
             ss.format = PA_SAMPLE_U8;
             break;
-        case AFMT_S16_LE:
+        case AF_FORMAT_S16_LE:
             ss.format = PA_SAMPLE_S16LE;
             break;
-        case AFMT_S16_BE:
+        case AF_FORMAT_S16_BE:
             ss.format = PA_SAMPLE_S16BE;
             break;
-        case AFMT_FLOAT:
+        case AF_FORMAT_FLOAT_NE:
             ss.format = PA_SAMPLE_FLOAT32;
             break;
         default:
diff --git a/libao2/ao_sdl.c b/libao2/ao_sdl.c
index 73a2c05a3d..bc2db2e258 100644
--- a/libao2/ao_sdl.c
+++ b/libao2/ao_sdl.c
@@ -20,7 +20,7 @@
 
 #include "audio_out.h"
 #include "audio_out_internal.h"
-#include "afmt.h"
+#include "libaf/af_format.h"
 #include <SDL.h>
 #include "osdep/timer.h"
 
@@ -181,7 +181,7 @@ static int init(int rate,int channels,int format,int flags){
 	/* Allocate ring-buffer memory */
 	buffer = (unsigned char *) malloc(BUFFSIZE);
 
-	mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_SDL_INFO, rate, (channels > 1) ? "Stereo" : "Mono", audio_out_format_name(format));
+//	mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_SDL_INFO, rate, (channels > 1) ? "Stereo" : "Mono", audio_out_format_name(format));
 
 	if(ao_subdevice) {
 		setenv("SDL_AUDIODRIVER", ao_subdevice, 1);
@@ -193,32 +193,32 @@ static int init(int rate,int channels,int format,int flags){
 	ao_data.format=format;
 
 	ao_data.bps=channels*rate;
-	if(format != AFMT_U8 && format != AFMT_S8)
+	if(format != AF_FORMAT_U8 && format != AF_FORMAT_S8)
 	  ao_data.bps*=2;
 	
 	/* The desired audio format (see SDL_AudioSpec) */
 	switch(format) {
-	    case AFMT_U8:
+	    case AF_FORMAT_U8:
 		aspec.format = AUDIO_U8;
 	    break;
-	    case AFMT_S16_LE:
+	    case AF_FORMAT_S16_LE:
 		aspec.format = AUDIO_S16LSB;
 	    break;
-	    case AFMT_S16_BE:
+	    case AF_FORMAT_S16_BE:
 		aspec.format = AUDIO_S16MSB;
 	    break;
-	    case AFMT_S8:
+	    case AF_FORMAT_S8:
 		aspec.format = AUDIO_S8;
 	    break;
-	    case AFMT_U16_LE:
+	    case AF_FORMAT_U16_LE:
 		aspec.format = AUDIO_U16LSB;
 	    break;
-	    case AFMT_U16_BE:
+	    case AF_FORMAT_U16_BE:
 		aspec.format = AUDIO_U16MSB;
 	    break;
 	    default:
                 aspec.format = AUDIO_S16LSB;
-                ao_data.format = AFMT_S16_LE;
+                ao_data.format = AF_FORMAT_S16_LE;
                 mp_msg(MSGT_AO,MSGL_WARN,MSGTR_AO_SDL_UnsupportedAudioFmt, format);
 	}
 
@@ -256,22 +256,22 @@ void callback(void *userdata, Uint8 *stream, int len); userdata is the pointer s
 
 	switch(obtained.format) {
 	    case AUDIO_U8 :
-		ao_data.format = AFMT_U8;
+		ao_data.format = AF_FORMAT_U8;
 	    break;
 	    case AUDIO_S16LSB :
-		ao_data.format = AFMT_S16_LE;
+		ao_data.format = AF_FORMAT_S16_LE;
 	    break;
 	    case AUDIO_S16MSB :
-		ao_data.format = AFMT_S16_BE;
+		ao_data.format = AF_FORMAT_S16_BE;
 	    break;
 	    case AUDIO_S8 :
-		ao_data.format = AFMT_S8;
+		ao_data.format = AF_FORMAT_S8;
 	    break;
 	    case AUDIO_U16LSB :
-		ao_data.format = AFMT_U16_LE;
+		ao_data.format = AF_FORMAT_U16_LE;
 	    break;
 	    case AUDIO_U16MSB :
-		ao_data.format = AFMT_U16_BE;
+		ao_data.format = AF_FORMAT_U16_BE;
 	    break;
 	    default:
                 mp_msg(MSGT_AO,MSGL_WARN,MSGTR_AO_SDL_UnsupportedAudioFmt, obtained.format);
diff --git a/libao2/ao_win32.c b/libao2/ao_win32.c
index ba004c7b26..e589ced8c1 100644
--- a/libao2/ao_win32.c
+++ b/libao2/ao_win32.c
@@ -23,7 +23,7 @@
 #include <windows.h>
 #include <mmsystem.h>
 
-#include "afmt.h"
+#include "libaf/af_format.h"
 #include "audio_out.h"
 #include "audio_out_internal.h"
 #include "mp_msg.h"
@@ -149,21 +149,21 @@ static int init(int rate,int channels,int format,int flags)
 	int i;
    
 	switch(format){
-		case AFMT_AC3:
-		case AFMT_S24_LE:
-		case AFMT_S16_LE:
-		case AFMT_S8:
+		case AF_FORMAT_AC3:
+		case AF_FORMAT_S24_LE:
+		case AF_FORMAT_S16_LE:
+		case AF_FORMAT_S8:
 			break;
 		default:
 			mp_msg(MSGT_AO, MSGL_V,"ao_win32: format %s not supported defaulting to Signed 16-bit Little-Endian\n",audio_out_format_name(format));
-			format=AFMT_S16_LE;
+			format=AF_FORMAT_S16_LE;
 	}   
 	//fill global ao_data 
 	ao_data.channels=channels;
 	ao_data.samplerate=rate;
 	ao_data.format=format;
 	ao_data.bps=channels*rate;
-	if(format != AFMT_U8 && format != AFMT_S8)
+	if(format != AF_FORMAT_U8 && format != AF_FORMAT_S8)
 	  ao_data.bps*=2;
 	if(ao_data.buffersize==-1)
 	{
@@ -179,7 +179,7 @@ static int init(int rate,int channels,int format,int flags)
     wformat.Format.cbSize          = (channels>2)?sizeof(WAVEFORMATEXTENSIBLE)-sizeof(WAVEFORMATEX):0;
     wformat.Format.nChannels       = channels;                
     wformat.Format.nSamplesPerSec  = rate;            
-    if(format == AFMT_AC3)
+    if(format == AF_FORMAT_AC3)
     {
         wformat.Format.wFormatTag      = WAVE_FORMAT_DOLBY_AC3_SPDIF;
         wformat.Format.wBitsPerSample  = 16;
@@ -208,7 +208,7 @@ static int init(int rate,int channels,int format,int flags)
 		mp_msg(MSGT_AO, MSGL_ERR,"ao_win32: format not supported switching to default\n");
         ao_data.channels = wformat.Format.nChannels = 2;
 	    ao_data.samplerate = wformat.Format.nSamplesPerSec = 44100;
-	    ao_data.format = AFMT_S16_LE;
+	    ao_data.format = AF_FORMAT_S16_LE;
 		ao_data.bps=ao_data.channels * ao_data.samplerate*2;
 	    wformat.Format.wBitsPerSample=16;
         wformat.Format.wFormatTag=WAVE_FORMAT_PCM;
diff --git a/libao2/audio_out.c b/libao2/audio_out.c
index 97691bc2d7..0bd37ad555 100644
--- a/libao2/audio_out.c
+++ b/libao2/audio_out.c
@@ -4,7 +4,6 @@
 
 #include "config.h"
 #include "audio_out.h"
-#include "afmt.h"
 
 #include "mp_msg.h"
 #include "help_mp.h"
@@ -187,6 +186,7 @@ ao_functions_t* init_best_audio_out(char** ao_list,int use_plugin,int rate,int c
 	    audio_out_plugin.control(AOCONTROL_SET_PLUGIN_DRIVER,audio_out);
 	    audio_out=&audio_out_plugin;
 	}
+//	if(audio_out->control(AOCONTROL_QUERY_FORMAT, (int)format) == CONTROL_TRUE)
 	if(audio_out->init(rate,channels,format,flags))
 	    return audio_out; // success!
     }
diff --git a/libao2/audio_plugin.h b/libao2/audio_plugin.h
index 170efea901..1936e19d36 100644
--- a/libao2/audio_plugin.h
+++ b/libao2/audio_plugin.h
@@ -45,7 +45,7 @@ extern ao_plugin_cfg_t ao_plugin_cfg;
 // Configuration defaults
 #define CFG_DEFAULTS { \
  NULL, \
- AFMT_S16_LE, \
+ AF_FORMAT_S16_LE, \
  0, \
  48000, \
  101, \
diff --git a/libao2/pl_delay.c b/libao2/pl_delay.c
index e39489505f..ed1693431c 100644
--- a/libao2/pl_delay.c
+++ b/libao2/pl_delay.c
@@ -12,7 +12,7 @@
 #include "audio_out.h"
 #include "audio_plugin.h"
 #include "audio_plugin_internal.h"
-#include "afmt.h"
+#include "libaf/af_format.h"
 
 static ao_info_t info =
 {
@@ -65,7 +65,7 @@ static int init(){
 
   // Tell ao_plugin how much this plugin adds to the overall time delay
   time_delay=-1*(float)ao_plugin_cfg.pl_delay_len/((float)pl_delay.channels*(float)pl_delay.rate);
-  if(pl_delay.format != AFMT_U8 && pl_delay.format != AFMT_S8)
+  if(pl_delay.format != AF_FORMAT_U8 && pl_delay.format != AF_FORMAT_S8)
     time_delay/=2;
   ao_plugin_data.delay_fix+=time_delay;
 
diff --git a/libao2/pl_eq.c b/libao2/pl_eq.c
index bb30470d91..1165b8033d 100644
--- a/libao2/pl_eq.c
+++ b/libao2/pl_eq.c
@@ -24,7 +24,7 @@
 #include "audio_out.h"
 #include "audio_plugin.h"
 #include "audio_plugin_internal.h"
-#include "afmt.h"
+#include "libaf/af_format.h"
 #include "eq.h"
 
 static ao_info_t info =
@@ -122,7 +122,7 @@ static int init(){
   float F[KM] = CF;
   
   // Check input format
-  if(ao_plugin_data.format != AFMT_S16_NE){
+  if(ao_plugin_data.format != AF_FORMAT_S16_NE){
     fprintf(stderr,"[pl_eq] Input audio format not yet supported. \n");
     return 0;
   }
diff --git a/libao2/pl_extrastereo.c b/libao2/pl_extrastereo.c
index 845a984a15..9f3404d5fe 100644
--- a/libao2/pl_extrastereo.c
+++ b/libao2/pl_extrastereo.c
@@ -19,7 +19,7 @@
 #include "audio_out.h"
 #include "audio_plugin.h"
 #include "audio_plugin_internal.h"
-#include "afmt.h"
+#include "libaf/af_format.h"
 
 static ao_info_t info = {
         "Extra stereo plugin",
@@ -57,7 +57,7 @@ static int control(int cmd,void *arg){
 // return: 1=success 0=fail
 static int init(){
   switch(ao_plugin_data.format){
-  case(AFMT_S16_NE):
+  case(AF_FORMAT_S16_NE):
     break;
   default:
     fprintf(stderr,"[pl_extrastereo] Audio format not yet suported \n");
@@ -87,7 +87,7 @@ static void reset(){
 static int play(){
 
   switch(pl_extrastereo.format){
-  case(AFMT_S16_NE): {
+  case(AF_FORMAT_S16_NE): {
 
     int16_t* data=(int16_t*)ao_plugin_data.data;
     int len=ao_plugin_data.len / 2; // 16 bits samples
diff --git a/libao2/pl_format.c b/libao2/pl_format.c
index 87b5bc8720..46ddaaa6a6 100644
--- a/libao2/pl_format.c
+++ b/libao2/pl_format.c
@@ -16,7 +16,7 @@
 #include "audio_out.h"
 #include "audio_plugin.h"
 #include "audio_plugin_internal.h"
-#include "afmt.h"
+#include "libaf/af_format.h"
 
 static ao_info_t info =
 {
@@ -75,29 +75,30 @@ static int control(int cmd,void *arg){
 // open & setup audio device
 // return: 1=success 0=fail
 static int init(){
+  char buf1[128], buf2[128];
   // Sheck input format
   switch(ao_plugin_data.format){
-  case(AFMT_U8):
+  case(AF_FORMAT_U8):
     pl_format.in=LE|B08|US; break;
-  case(AFMT_S8):
+  case(AF_FORMAT_S8):
     pl_format.in=LE|B08|SI; break;
-  case(AFMT_S16_LE):
+  case(AF_FORMAT_S16_LE):
     pl_format.in=LE|B16|SI; break;
-  case(AFMT_S16_BE):
+  case(AF_FORMAT_S16_BE):
     pl_format.in=BE|B16|SI; break;
-  case(AFMT_U16_LE):	
+  case(AF_FORMAT_U16_LE):	
     pl_format.in=LE|B16|US; break;
-  case(AFMT_U16_BE):	
+  case(AF_FORMAT_U16_BE):	
     pl_format.in=BE|B16|US; break;
-  case(AFMT_S32_LE):
+  case(AF_FORMAT_S32_LE):
     pl_format.in=LE|B32|SI; break;
-  case(AFMT_S32_BE):	
+  case(AF_FORMAT_S32_BE):	
     pl_format.in=BE|B32|SI; break;
-  case(AFMT_IMA_ADPCM):		
-  case(AFMT_MU_LAW):
-  case(AFMT_A_LAW):
-  case(AFMT_MPEG):
-  case(AFMT_AC3):
+  case(AF_FORMAT_IMA_ADPCM):		
+  case(AF_FORMAT_MU_LAW):
+  case(AF_FORMAT_A_LAW):
+  case(AF_FORMAT_MPEG2):
+  case(AF_FORMAT_AC3):
     printf("[pl_format] Input audio format not yet suported \n");
     return 0;
   default: 
@@ -106,27 +107,27 @@ static int init(){
   }
   // Sheck output format
   switch(ao_plugin_cfg.pl_format_type){
-  case(AFMT_U8):
+  case(AF_FORMAT_U8):
     pl_format.out=LE|B08|US; break;
-  case(AFMT_S8):
+  case(AF_FORMAT_S8):
     pl_format.out=LE|B08|SI; break;
-  case(AFMT_S16_LE):
+  case(AF_FORMAT_S16_LE):
     pl_format.out=LE|B16|SI; break;
-  case(AFMT_S16_BE):
+  case(AF_FORMAT_S16_BE):
     pl_format.out=BE|B16|SI; break;
-  case(AFMT_U16_LE):	
+  case(AF_FORMAT_U16_LE):	
     pl_format.out=LE|B16|US; break;
-  case(AFMT_U16_BE):	
+  case(AF_FORMAT_U16_BE):	
     pl_format.out=BE|B16|US; break;
-  case(AFMT_S32_LE):
+  case(AF_FORMAT_S32_LE):
     pl_format.out=LE|B32|SI; break;
-  case(AFMT_S32_BE):	
+  case(AF_FORMAT_S32_BE):	
     pl_format.out=BE|B32|SI; break;
-  case(AFMT_IMA_ADPCM):		
-  case(AFMT_MU_LAW):
-  case(AFMT_A_LAW):
-  case(AFMT_MPEG):
-  case(AFMT_AC3):
+  case(AF_FORMAT_IMA_ADPCM):		
+  case(AF_FORMAT_MU_LAW):
+  case(AF_FORMAT_A_LAW):
+  case(AF_FORMAT_MPEG2):
+  case(AF_FORMAT_AC3):
     printf("[pl_format] Output audio format not yet suported \n");
     return 0;
   default:
@@ -136,8 +137,8 @@ static int init(){
 
   // Tell the world what we are up to
   printf("[pl_format] Input format: %s, output format: %s \n",
-	 audio_out_format_name(ao_plugin_data.format),
-	 audio_out_format_name(ao_plugin_cfg.pl_format_type));
+	 af_fmt2str(ao_plugin_data.format, &buf1, 128),
+	 af_fmt2str(ao_plugin_cfg.pl_format_type, &buf2, 128));
 
   // We are changing the format
   ao_plugin_data.format=ao_plugin_cfg.pl_format_type;
diff --git a/libao2/pl_resample.c b/libao2/pl_resample.c
index 505eb0aa1d..b240fdd25e 100644
--- a/libao2/pl_resample.c
+++ b/libao2/pl_resample.c
@@ -23,7 +23,7 @@
 #include "audio_out.h"
 #include "audio_plugin.h"
 #include "audio_plugin_internal.h"
-#include "afmt.h"
+#include "libaf/af_format.h"
 
 static ao_info_t info =
 {
@@ -119,7 +119,7 @@ static int init(){
   pl_resample.up=UP;
 
   // Sheck input format
-  if(ao_plugin_data.format != AFMT_S16_NE){
+  if(ao_plugin_data.format != AF_FORMAT_S16_NE){
     fprintf(stderr,"[pl_resample] Input audio format not yet suported. \n");
     return 0;
   }
diff --git a/libao2/pl_surround.c b/libao2/pl_surround.c
index e0e6e52570..6700bd5471 100644
--- a/libao2/pl_surround.c
+++ b/libao2/pl_surround.c
@@ -44,7 +44,7 @@
 #include "audio_out.h"
 #include "audio_plugin.h"
 #include "audio_plugin_internal.h"
-#include "afmt.h"
+#include "libaf/af_format.h"
 
 #include "remez.h"
 #include "firfilter.c"
@@ -108,8 +108,8 @@ static int init(){
     pl_surround.passthrough = 1;
     return 1;
   }
-  if (ao_plugin_data.format != AFMT_S16_NE) {
-    fprintf(stderr, "pl_surround: I'm dumb and can only handle AFMT_S16_NE audio format, using passthrough mode\n");
+  if (ao_plugin_data.format != AF_FORMAT_S16_NE) {
+    fprintf(stderr, "pl_surround: I'm dumb and can only handle AF_FORMAT_S16_NE audio format, using passthrough mode\n");
     pl_surround.passthrough = 1;
     return 1;
   }
diff --git a/libao2/pl_volnorm.c b/libao2/pl_volnorm.c
index dd5d0a4e9b..4a4823a244 100644
--- a/libao2/pl_volnorm.c
+++ b/libao2/pl_volnorm.c
@@ -30,7 +30,7 @@
 #include "audio_out.h"
 #include "audio_plugin.h"
 #include "audio_plugin_internal.h"
-#include "afmt.h"
+#include "libaf/af_format.h"
 
 static ao_info_t info = {
         "Volume normalizer",
@@ -116,7 +116,7 @@ static int control(int cmd,void *arg){
 // return: 1=success 0=fail
 static int init(){
   switch(ao_plugin_data.format){
-    case(AFMT_S16_NE):
+    case(AF_FORMAT_S16_NE):
       break;
     default:
       fprintf(stderr,"[pl_volnorm] Audio format not yet supported.\n");
@@ -142,7 +142,7 @@ static void reset(){
   int i;
   mul = MUL_INIT;
   switch(ao_plugin_data.format) {
-    case(AFMT_S16_NE):
+    case(AF_FORMAT_S16_NE):
 #if AVG==1
       lastavg = MID_S16;
 #elif AVG==2
@@ -165,7 +165,7 @@ static void reset(){
 static int play(){
 
   switch(pl_volnorm.format){
-  case(AFMT_S16_NE): {
+  case(AF_FORMAT_S16_NE): {
 #define CLAMP(x,m,M) do { if ((x)<(m)) (x) = (m); else if ((x)>(M)) (x) = (M); } while(0)
 
     int16_t* data=(int16_t*)ao_plugin_data.data;
diff --git a/libao2/pl_volume.c b/libao2/pl_volume.c
index bce069a8c4..2b673d3e1e 100644
--- a/libao2/pl_volume.c
+++ b/libao2/pl_volume.c
@@ -20,7 +20,7 @@
 #include "audio_out.h"
 #include "audio_plugin.h"
 #include "audio_plugin_internal.h"
-#include "afmt.h"
+#include "libaf/af_format.h"
 
 static ao_info_t info =
 {
@@ -80,8 +80,8 @@ static int control(int cmd,void *arg){
 static int init(){
   // Sanity sheck this plugin supports AFMT_U8 and AFMT_S16_LE
   switch(ao_plugin_data.format){
-  case(AFMT_U8):
-  case(AFMT_S16_NE):
+  case(AF_FORMAT_U8):
+  case(AF_FORMAT_S16_NE):
     break;
   default: 
     fprintf(stderr,"[pl_volume] Audio format not yet suported \n");
@@ -117,7 +117,7 @@ static int play(){
   vol=(vol*vol*vol)>>12;
   // Change the volume.
   switch(pl_volume.format){
-  case(AFMT_U8):{
+  case(AF_FORMAT_U8):{
     register uint8_t* data=(uint8_t*)ao_plugin_data.data;
     register int x;
       for(i=0;i<ao_plugin_data.len;i++){
@@ -135,7 +135,7 @@ static int play(){
       }
       break;
   }
-  case(AFMT_S16_NE):{
+  case(AF_FORMAT_S16_NE):{
     register int len=ao_plugin_data.len>>1;
     register int16_t* data=(int16_t*)ao_plugin_data.data;
     register int x;