hopefully final fix

git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@14250 b3059339-0415-0410-9bf9-f77b7e298cf2

6 files changed, 21 insertions, 20 deletions

diff --git a/libao2/ao_alsa.c b/libao2/ao_alsa.c
index 539fd0fa7c..db23f1b532 100644
--- a/libao2/ao_alsa.c
+++ b/libao2/ao_alsa.c
@@ -256,8 +256,8 @@ static int init(int rate_hz, int channels, int format, int flags)
     // make sure alsa_device is null-terminated even when using strncpy etc.
     memset(alsa_device, 0, ALSA_DEVICE_SIZE + 1);
 
-    mp_msg(MSGT_AO,MSGL_V,"alsa-init: requested format: %d Hz, %d channels, %s\n", rate_hz,
-	channels, audio_out_format_name(format));
+    mp_msg(MSGT_AO,MSGL_V,"alsa-init: requested format: %d Hz, %d channels, %x\n", rate_hz,
+	channels, format);
     alsa_handler = NULL;
     mp_msg(MSGT_AO,MSGL_V,"alsa-init: compiled for ALSA-%s\n", SND_LIB_VERSION_STR);
     
@@ -334,8 +334,7 @@ static int init(int rate_hz, int channels, int format, int flags)
 	ao_data.bps *= 4;
 	break;
       case -1:
-	mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: invalid format (%s) requested - output disabled\n",
-	       audio_out_format_name(format));
+	mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: invalid format (%x) requested - output disabled\n",format);
 	return(0);
 	break;
       default:
@@ -587,8 +586,7 @@ static int init(int rate_hz, int channels, int format, int flags)
                                              alsa_format)) < 0)
       {
          mp_msg(MSGT_AO,MSGL_INFO,
-		"alsa-init: format %s are not supported by hardware, trying default\n", 
-		audio_out_format_name(format));
+		"alsa-init: format %x are not supported by hardware, trying default\n", format);
          alsa_format = SND_PCM_FORMAT_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 ed073f55ae..687b48ac8c 100644
--- a/libao2/ao_alsa5.c
+++ b/libao2/ao_alsa5.c
@@ -50,9 +50,10 @@ static int init(int rate_hz, int channels, int format, int flags)
     snd_pcm_channel_setup_t setup;
     snd_pcm_info_t info;
     snd_pcm_channel_info_t chninfo;
+    char buf[128];
 
     mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_ALSA5_InitInfo, rate_hz,
-	channels, audio_out_format_name(format));
+	channels, af_fmt2str(format));
 
     alsa_handler = NULL;
 
@@ -111,8 +112,7 @@ static int init(int rate_hz, int channels, int format, int flags)
 	    ao_data.bps *= 2;
 	    break;
 	case -1:
-	    mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_InvalidFormatReq,
-		audio_out_format_name(format));
+	    mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_InvalidFormatReq,af_fmt2str(format,&buf,128));
 	    return(0);
 	default:
 	    break;
diff --git a/libao2/ao_dsound.c b/libao2/ao_dsound.c
index 39ab901df8..bcdce0b1a0 100644
--- a/libao2/ao_dsound.c
+++ b/libao2/ao_dsound.c
@@ -274,7 +274,7 @@ static int write_buffer(unsigned char *data, int len)
   	    int i, j;
   	    int numsamp,sampsize;
 
-  	    sampsize = audio_out_format_bits(ao_data.format)>>3; // bytes per sample
+  	    sampsize = af_fmt2bits(ao_data.format)>>3; // bytes per sample
   	    numsamp = dwBytes1 / (ao_data.channels * sampsize);  // number of samples for each channel in this buffer
 
   	    for( i = 0; i < numsamp; i++ ) for( j = 0; j < ao_data.channels; j++ ) {
@@ -372,16 +372,16 @@ static int init(int rate, int channels, int format, int flags)
 		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));
+			mp_msg(MSGT_AO, MSGL_V,"ao_dsound: format %x not supported defaulting to Signed 16-bit Little-Endian\n",format);
 			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 * (audio_out_format_bits(format)>>3);
+	ao_data.bps = channels * rate * (af_fmt2bits(format)>>3);
 	if(ao_data.buffersize==-1) ao_data.buffersize = ao_data.bps; // space for 1 sec
-	mp_msg(MSGT_AO, MSGL_V,"ao_dsound: Samplerate:%iHz Channels:%i Format:%s\n", rate, channels, audio_out_format_name(format));
+	mp_msg(MSGT_AO, MSGL_V,"ao_dsound: Samplerate:%iHz Channels:%i Format:%x\n", rate, channels, format);
 	mp_msg(MSGT_AO, MSGL_V,"ao_dsound: Buffersize:%d bytes (%d msec)\n", ao_data.buffersize, ao_data.buffersize / ao_data.bps * 1000);
 
 	//fill waveformatex
@@ -395,7 +395,7 @@ static int init(int rate, int channels, int format, int flags)
 		wformat.Format.nBlockAlign     = 4;
 	} else {
 		wformat.Format.wFormatTag      = (channels > 2) ? WAVE_FORMAT_EXTENSIBLE : WAVE_FORMAT_PCM;
-		wformat.Format.wBitsPerSample  = audio_out_format_bits(format);
+		wformat.Format.wBitsPerSample  = af_fmt2bits(format);
 		wformat.Format.nBlockAlign     = wformat.Format.nChannels * (wformat.Format.wBitsPerSample >> 3);
 	}
 
diff --git a/libao2/ao_sgi.c b/libao2/ao_sgi.c
index d1c6ef2d1d..796691ab4e 100644
--- a/libao2/ao_sgi.c
+++ b/libao2/ao_sgi.c
@@ -41,8 +41,9 @@ static int control(int cmd, void *arg){
 // open & setup audio device
 // return: 1=success 0=fail
 static int init(int rate, int channels, int format, int flags) {
-  
-  mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_InitInfo, rate, (channels > 1) ? "Stereo" : "Mono", audio_out_format_name(format));
+
+  char buf[128];  
+  mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_InitInfo, rate, (channels > 1) ? "Stereo" : "Mono", af_fmt2str(format, &buf, 128));
   
   { /* from /usr/share/src/dmedia/audio/setrate.c */
   
diff --git a/libao2/ao_sun.c b/libao2/ao_sun.c
index b9b6a3a9f4..2e102ccfa1 100644
--- a/libao2/ao_sun.c
+++ b/libao2/ao_sun.c
@@ -457,6 +457,7 @@ static int init(int rate,int channels,int format,int flags){
     audio_info_t info;
     int pass;
     int ok;
+    char buf[128];
 
     setup_device_paths();
 
@@ -479,7 +480,7 @@ static int init(int rate,int channels,int format,int flags){
     for (ok = pass = 0; pass <= 5; pass++) { /* pass 6&7 not useful */
 
 	AUDIO_INITINFO(&info);
-	info.play.encoding = oss2sunfmt(ao_data.format = format);
+	info.play.encoding = af2sunfmt(ao_data.format = format);
 	info.play.precision =
 	    (format==AF_FORMAT_S16_LE || format==AF_FORMAT_S16_BE
 	     ? AUDIO_PRECISION_16
@@ -545,7 +546,7 @@ static int init(int rate,int channels,int format,int flags){
 
     if (!ok) {
 	mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SUN_UnsupSampleRate,
-	       channels, audio_out_format_name(format), rate);
+	       channels, af_fmt2str(format, &buf, 128), rate);
 	return 0;
     }
 
@@ -625,7 +626,7 @@ static void reset(){
     ioctl(audio_fd, AUDIO_DRAIN, 0);
 
     AUDIO_INITINFO(&info);
-    info.play.encoding = oss2sunfmt(ao_data.format);
+    info.play.encoding = af2sunfmt(ao_data.format);
     info.play.precision =
 	(ao_data.format==AF_FORMAT_S16_LE || ao_data.format==AF_FORMAT_S16_BE 
 	 ? AUDIO_PRECISION_16
diff --git a/libao2/ao_win32.c b/libao2/ao_win32.c
index e589ced8c1..ff07a9e73c 100644
--- a/libao2/ao_win32.c
+++ b/libao2/ao_win32.c
@@ -147,6 +147,7 @@ static int init(int rate,int channels,int format,int flags)
 	MMRESULT result;
 	unsigned char* buffer;
 	int i;
+	char buf[128];
    
 	switch(format){
 		case AF_FORMAT_AC3:
@@ -155,7 +156,7 @@ static int init(int rate,int channels,int format,int flags)
 		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));
+			mp_msg(MSGT_AO, MSGL_V,"ao_win32: format %s not supported defaulting to Signed 16-bit Little-Endian\n",af_fmt2str(format, &buf, 128));
 			format=AF_FORMAT_S16_LE;
 	}   
 	//fill global ao_data