interface to libao2 changed ao_plugin added

git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@3096 b3059339-0415-0410-9bf9-f77b7e298cf2

14 files changed, 204 insertions, 265 deletions

diff --git a/libao2/Makefile b/libao2/Makefile
index b0b8d2a96d..beb4cff8d3 100644
--- a/libao2/Makefile
+++ b/libao2/Makefile
@@ -4,7 +4,7 @@ include config.mak
 LIBNAME = libao2.a
 
 # TODO: moveout ao_sdl.c so it's only used when SDL is detected
-SRCS=afmt.c audio_out.c ao_null.c ao_pcm.c ao_mpegpes.c $(OPTIONAL_SRCS)
+SRCS=afmt.c audio_out.c ao_mpegpes.c ao_null.c ao_pcm.c ao_plugin.c $(OPTIONAL_SRCS)
 OBJS=$(SRCS:.c=.o)
 
 CFLAGS  = $(OPTFLAGS) -I. -I.. $(SDL_INC) $(EXTRA_INC)
@@ -38,3 +38,4 @@ depend:
 ifneq ($(wildcard .depend),)
 include .depend
 endif
+
diff --git a/libao2/ao_alsa1x.c b/libao2/ao_alsa1x.c
index bebc04a0fe..de05131fda 100644
--- a/libao2/ao_alsa1x.c
+++ b/libao2/ao_alsa1x.c
@@ -32,14 +32,6 @@ static ao_info_t info =
 
 LIBAO_EXTERN(alsa9)
 
-/* global variables:
-    ao_samplerate
-    ao_channels
-    ao_format
-    ao_bps
-    ao_outburst
-    ao_buffersize
-*/
 
 static snd_pcm_t *alsa_handler;
 static snd_pcm_format_t alsa_format;
@@ -67,7 +59,7 @@ static int control(int cmd, int arg)
 
 	    strncpy(alsa_device, (char *)arg, ALSA_DEVICE_SIZE);
 	    uninit();
-	    ret = init(ao_samplerate, ao_channels, ao_format, 0);
+	    ret = init(ao_data.samplerate, ao_data.channels, ao_data.format, 0);
 	    if (ret == 0)
 		return(CONTROL_ERROR);
 	    else
@@ -112,12 +104,12 @@ static int init(int rate_hz, int channels, int format, int flags)
 	return(0);
     }
 
-    ao_samplerate = rate_hz;
-    ao_bps = channels; /* really this is bytes per frame so bad varname */
-    ao_format = format;
-    ao_channels = channels;
-    ao_outburst = OUTBURST;
-    ao_buffersize = 16384;
+    ao_data.samplerate = rate_hz;
+    ao_data.bps = channels; /* really this is bytes per frame so bad varname */
+    ao_data.format = format;
+    ao_data.channels = channels;
+    ao_data.outburst = OUTBURST;
+    ao_data.buffersize = 16384;
 
     switch (format)
     {
@@ -148,7 +140,7 @@ static int init(int rate_hz, int channels, int format, int flags)
     {
 	case SND_PCM_FORMAT_S16_LE:
 	case SND_PCM_FORMAT_U16_LE:
-	    ao_bps *= 2;
+	    ao_data.bps *= 2;
 	    break;
 	case -1:
 	    printf("alsa-init: invalid format (%s) requested - output disabled\n",
@@ -164,8 +156,8 @@ static int init(int rate_hz, int channels, int format, int flags)
 	return(0);
     }
 
-    if (ao_subdevice != NULL)
-	alsa_device = ao_subdevice;
+    if (ao_data.subdevice != NULL)
+	alsa_device = ao_data.subdevice;
 
     if (alsa_device == NULL)
     {
@@ -219,14 +211,14 @@ static int init(int rate_hz, int channels, int format, int flags)
     }
 
     if ((err = snd_pcm_hw_params_set_channels(alsa_handler, alsa_hwparams,
-	ao_channels)) < 0)
+	ao_data.channels)) < 0)
     {
 	printf("alsa-init: unable to set channels: %s\n",
 	    snd_strerror(err));
 	return(0);
     }
 
-        if ((err = snd_pcm_hw_params_set_rate_near(alsa_handler, alsa_hwparams, ao_samplerate, 0)) < 0) 
+        if ((err = snd_pcm_hw_params_set_rate_near(alsa_handler, alsa_hwparams, ao_data.samplerate, 0)) < 0) 
 /* was originally only snd_pcm_hw_params_set_rate jp*/ 
         {
     	printf("alsa-init: unable to set samplerate-2: %s\n",
@@ -259,9 +251,9 @@ static int init(int rate_hz, int channels, int format, int flags)
 	return(0);
     } else
     {
-	ao_buffersize = err;
+	ao_data.buffersize = err;
         if (verbose)
-	    printf("alsa-init: got buffersize %i\n", ao_buffersize);
+	    printf("alsa-init: got buffersize %i\n", ao_data.buffersize);
     }
 #endif
 
@@ -278,7 +270,7 @@ static int init(int rate_hz, int channels, int format, int flags)
 	    alsa_buffer_time = err;
 
 	if ((err = snd_pcm_hw_params_set_period_time_near(alsa_handler, alsa_hwparams, alsa_buffer_time/4, 0)) < 0)
-	  /* original: alsa_buffer_time/ao_bps */
+	  /* original: alsa_buffer_time/ao_data.bps */
 	{
 	    printf("alsa-init: unable to set period time: %s\n",
 		snd_strerror(err));
@@ -299,7 +291,7 @@ static int init(int rate_hz, int channels, int format, int flags)
 #ifdef sw_params
     {
     chunk_size = snd_pcm_hw_params_get_period_size(alsa_hwparams, 0);
-    start_threshold = (double) ao_samplerate * start_delay / 1000000;
+    start_threshold = (double) ao_data.samplerate * start_delay / 1000000;
     xfer_align = snd_pcm_sw_params_get_xfer_align(alsa_swparams);
 
     if ((err = snd_pcm_sw_params_current(alsa_handler, alsa_swparams)) < 0)
@@ -369,7 +361,7 @@ static int init(int rate_hz, int channels, int format, int flags)
     }
 #endif
     printf("AUDIO: %d Hz/%d channels/%d bpf/%d bytes buffer/%s\n",
-	ao_samplerate, ao_channels, ao_bps, ao_buffersize,
+	ao_data.samplerate, ao_data.channels, ao_data.bps, ao_data.buffersize,
 	snd_pcm_format_description(alsa_format));
     return(1);
 }
@@ -488,7 +480,7 @@ static int play(void* data, int len, int flags)
 
     got_len = snd_pcm_writei(alsa_handler, data, len / 4);
     
-    //if ((got_len = snd_pcm_writei(alsa_handler, data, (len/ao_bps))) != (len/ao_bps)) {     
+    //if ((got_len = snd_pcm_writei(alsa_handler, data, (len/ao_data.bps))) != (len/ao_data.bps)) {     
     //SHOULD BE FIXED      
 	if (got_len == -EPIPE) /* underrun? */
 	{
@@ -498,7 +490,7 @@ static int play(void* data, int len, int flags)
 		printf("alsa-play: playback prepare error: %s\n", snd_strerror(got_len));
 		return(0);
 	    }
-	    if ((got_len = snd_pcm_writei(alsa_handler, data, (len/ao_bps))) != (len/ao_bps))
+	    if ((got_len = snd_pcm_writei(alsa_handler, data, (len/ao_data.bps))) != (len/ao_data.bps))
 	    {
 		printf("alsa-play: write error after reset: %s - giving up\n",
 		    snd_strerror(got_len));
@@ -533,7 +525,7 @@ static int get_space()
 	case SND_PCM_STATE_OPEN:
 	case SND_PCM_STATE_PREPARED:
 	case SND_PCM_STATE_RUNNING:
-	    ret = snd_pcm_status_get_avail(status) * ao_bps;
+	    ret = snd_pcm_status_get_avail(status) * ao_data.bps;
 	    break;
 	default:
 	    ret = 0;
@@ -546,11 +538,11 @@ static int get_space()
     return(ret);
 }
 
-/* how many unplayed bytes are in the buffer */
-static int get_delay()
+/* delay in seconds between first and last sample in buffer */
+static float get_delay()
 {
     snd_pcm_status_t *status;
-    int ret;
+    float ret;
     
     if ((ret = snd_pcm_status_malloc(&status)) < 0)
     {
@@ -569,7 +561,7 @@ static int get_delay()
 	case SND_PCM_STATE_OPEN:
 	case SND_PCM_STATE_PREPARED:
 	case SND_PCM_STATE_RUNNING:
-	    ret = snd_pcm_status_get_delay(status) * ao_bps;
+	    ret = (float)snd_pcm_status_get_delay(status)/(float)ao_data.samplerate;
 	    break;
 	default:
 	    ret = 0;
diff --git a/libao2/ao_alsa5.c b/libao2/ao_alsa5.c
index fd037c0f03..a7c12cba43 100644
--- a/libao2/ao_alsa5.c
+++ b/libao2/ao_alsa5.c
@@ -27,15 +27,6 @@ static ao_info_t info =
 
 LIBAO_EXTERN(alsa5)
 
-/* global variables:
-    ao_samplerate
-    ao_channels
-    ao_format
-    ao_bps
-    ao_outburst
-    ao_buffersize
-*/
-
 static snd_pcm_t *alsa_handler;
 static snd_pcm_format_t alsa_format;
 static int alsa_rate = SND_PCM_RATE_CONTINUOUS;
@@ -74,12 +65,12 @@ static int init(int rate_hz, int channels, int format, int flags)
 	return(0);
     }
 
-    ao_format = format;
-    ao_channels = channels - 1;
-    ao_samplerate = rate_hz;
-    ao_bps = ao_samplerate*(ao_channels+1);
-    ao_outburst = OUTBURST;
-    ao_buffersize = 16384;
+    ao_data.format = format;
+    ao_data.channels = channels - 1;
+    ao_data.samplerate = rate_hz;
+    ao_data.bps = ao_data.samplerate*(ao_data.channels+1);
+    ao_data.outburst = OUTBURST;
+    ao_data.buffersize = 16384;
 
     memset(&alsa_format, 0, sizeof(alsa_format));
     switch (format)
@@ -111,7 +102,7 @@ static int init(int rate_hz, int channels, int format, int flags)
     {
 	case SND_PCM_SFMT_S16_LE:
 	case SND_PCM_SFMT_U16_LE:
-	    ao_bps *= 2;
+	    ao_data.bps *= 2;
 	    break;
 	case -1:
 	    printf("alsa-init: invalid format (%s) requested - output disabled\n",
@@ -161,8 +152,8 @@ static int init(int rate_hz, int channels, int format, int flags)
 	    break;
     }
 
-    alsa_format.rate = ao_samplerate;
-    alsa_format.voices = ao_channels*2;
+    alsa_format.rate = ao_data.samplerate;
+    alsa_format.voices = ao_data.channels*2;
     alsa_format.interleave = 1;
 
     if ((err = snd_pcm_open(&alsa_handler, 0, 0, SND_PCM_OPEN_PLAYBACK)) < 0)
@@ -189,13 +180,15 @@ static int init(int rate_hz, int channels, int format, int flags)
 	    printf("alsa-init: pcm channel info error: %s\n", snd_strerror(err));
 	    return(0);
 	}
+
 #ifndef __QNX__
 	if (chninfo.buffer_size)
-	    ao_buffersize = chninfo.buffer_size;
+	    ao_data.buffersize = chninfo.buffer_size;
 #endif
+
 	if (verbose)
 	    printf("alsa-init: setting preferred buffer size from driver: %d bytes\n",
-		ao_buffersize);
+		ao_data.buffersize);
     }
 
     memset(&params, 0, sizeof(params));
@@ -204,7 +197,7 @@ static int init(int rate_hz, int channels, int format, int flags)
     params.format = alsa_format;
     params.start_mode = SND_PCM_START_DATA;
     params.stop_mode = SND_PCM_STOP_ROLLOVER;
-    params.buf.stream.queue_size = ao_buffersize;
+    params.buf.stream.queue_size = ao_data.buffersize;
     params.buf.stream.fill = SND_PCM_FILL_NONE;
 
     if ((err = snd_pcm_channel_params(alsa_handler, &params)) < 0)
@@ -217,8 +210,8 @@ static int init(int rate_hz, int channels, int format, int flags)
     setup.channel = SND_PCM_CHANNEL_PLAYBACK;
     setup.mode = SND_PCM_MODE_STREAM;
     setup.format = alsa_format;
-    setup.buf.stream.queue_size = ao_buffersize;
-    setup.msbits_per_sample = ao_bps;
+    setup.buf.stream.queue_size = ao_data.buffersize;
+    setup.msbits_per_sample = ao_data.bps;
     
     if ((err = snd_pcm_channel_setup(alsa_handler, &setup)) < 0)
     {
@@ -233,7 +226,7 @@ static int init(int rate_hz, int channels, int format, int flags)
     }
 
     printf("AUDIO: %d Hz/%d channels/%d bps/%d bytes buffer/%s\n",
-	ao_samplerate, ao_channels+1, ao_bps, ao_buffersize,
+	ao_data.samplerate, ao_data.channels+1, ao_data.bps, ao_data.buffersize,
 	snd_pcm_get_format_name(alsa_format.format));
     return(1);
 }
@@ -357,15 +350,15 @@ static int get_space()
 	return(ch_stat.free);
 }
 
-/* how many unplayed bytes are in the buffer */
-static int get_delay()
+/* delay in seconds between first and last sample in buffer */
+static float get_delay()
 {
     snd_pcm_channel_status_t ch_stat;
     
     ch_stat.channel = SND_PCM_CHANNEL_PLAYBACK;
     
     if (snd_pcm_channel_status(alsa_handler, &ch_stat) < 0)
-	return(ao_buffersize); /* error occured */
+	return((float)ao_data.buffersize/(float)ao_data.bps); /* error occured */
     else
-	return(ch_stat.count);
+	return((float)ch_stat.count/(float)ao_data.bps);
 }
diff --git a/libao2/ao_alsa9.c b/libao2/ao_alsa9.c
index bebc04a0fe..de05131fda 100644
--- a/libao2/ao_alsa9.c
+++ b/libao2/ao_alsa9.c
@@ -32,14 +32,6 @@ static ao_info_t info =
 
 LIBAO_EXTERN(alsa9)
 
-/* global variables:
-    ao_samplerate
-    ao_channels
-    ao_format
-    ao_bps
-    ao_outburst
-    ao_buffersize
-*/
 
 static snd_pcm_t *alsa_handler;
 static snd_pcm_format_t alsa_format;
@@ -67,7 +59,7 @@ static int control(int cmd, int arg)
 
 	    strncpy(alsa_device, (char *)arg, ALSA_DEVICE_SIZE);
 	    uninit();
-	    ret = init(ao_samplerate, ao_channels, ao_format, 0);
+	    ret = init(ao_data.samplerate, ao_data.channels, ao_data.format, 0);
 	    if (ret == 0)
 		return(CONTROL_ERROR);
 	    else
@@ -112,12 +104,12 @@ static int init(int rate_hz, int channels, int format, int flags)
 	return(0);
     }
 
-    ao_samplerate = rate_hz;
-    ao_bps = channels; /* really this is bytes per frame so bad varname */
-    ao_format = format;
-    ao_channels = channels;
-    ao_outburst = OUTBURST;
-    ao_buffersize = 16384;
+    ao_data.samplerate = rate_hz;
+    ao_data.bps = channels; /* really this is bytes per frame so bad varname */
+    ao_data.format = format;
+    ao_data.channels = channels;
+    ao_data.outburst = OUTBURST;
+    ao_data.buffersize = 16384;
 
     switch (format)
     {
@@ -148,7 +140,7 @@ static int init(int rate_hz, int channels, int format, int flags)
     {
 	case SND_PCM_FORMAT_S16_LE:
 	case SND_PCM_FORMAT_U16_LE:
-	    ao_bps *= 2;
+	    ao_data.bps *= 2;
 	    break;
 	case -1:
 	    printf("alsa-init: invalid format (%s) requested - output disabled\n",
@@ -164,8 +156,8 @@ static int init(int rate_hz, int channels, int format, int flags)
 	return(0);
     }
 
-    if (ao_subdevice != NULL)
-	alsa_device = ao_subdevice;
+    if (ao_data.subdevice != NULL)
+	alsa_device = ao_data.subdevice;
 
     if (alsa_device == NULL)
     {
@@ -219,14 +211,14 @@ static int init(int rate_hz, int channels, int format, int flags)
     }
 
     if ((err = snd_pcm_hw_params_set_channels(alsa_handler, alsa_hwparams,
-	ao_channels)) < 0)
+	ao_data.channels)) < 0)
     {
 	printf("alsa-init: unable to set channels: %s\n",
 	    snd_strerror(err));
 	return(0);
     }
 
-        if ((err = snd_pcm_hw_params_set_rate_near(alsa_handler, alsa_hwparams, ao_samplerate, 0)) < 0) 
+        if ((err = snd_pcm_hw_params_set_rate_near(alsa_handler, alsa_hwparams, ao_data.samplerate, 0)) < 0) 
 /* was originally only snd_pcm_hw_params_set_rate jp*/ 
         {
     	printf("alsa-init: unable to set samplerate-2: %s\n",
@@ -259,9 +251,9 @@ static int init(int rate_hz, int channels, int format, int flags)
 	return(0);
     } else
     {
-	ao_buffersize = err;
+	ao_data.buffersize = err;
         if (verbose)
-	    printf("alsa-init: got buffersize %i\n", ao_buffersize);
+	    printf("alsa-init: got buffersize %i\n", ao_data.buffersize);
     }
 #endif
 
@@ -278,7 +270,7 @@ static int init(int rate_hz, int channels, int format, int flags)
 	    alsa_buffer_time = err;
 
 	if ((err = snd_pcm_hw_params_set_period_time_near(alsa_handler, alsa_hwparams, alsa_buffer_time/4, 0)) < 0)
-	  /* original: alsa_buffer_time/ao_bps */
+	  /* original: alsa_buffer_time/ao_data.bps */
 	{
 	    printf("alsa-init: unable to set period time: %s\n",
 		snd_strerror(err));
@@ -299,7 +291,7 @@ static int init(int rate_hz, int channels, int format, int flags)
 #ifdef sw_params
     {
     chunk_size = snd_pcm_hw_params_get_period_size(alsa_hwparams, 0);
-    start_threshold = (double) ao_samplerate * start_delay / 1000000;
+    start_threshold = (double) ao_data.samplerate * start_delay / 1000000;
     xfer_align = snd_pcm_sw_params_get_xfer_align(alsa_swparams);
 
     if ((err = snd_pcm_sw_params_current(alsa_handler, alsa_swparams)) < 0)
@@ -369,7 +361,7 @@ static int init(int rate_hz, int channels, int format, int flags)
     }
 #endif
     printf("AUDIO: %d Hz/%d channels/%d bpf/%d bytes buffer/%s\n",
-	ao_samplerate, ao_channels, ao_bps, ao_buffersize,
+	ao_data.samplerate, ao_data.channels, ao_data.bps, ao_data.buffersize,
 	snd_pcm_format_description(alsa_format));
     return(1);
 }
@@ -488,7 +480,7 @@ static int play(void* data, int len, int flags)
 
     got_len = snd_pcm_writei(alsa_handler, data, len / 4);
     
-    //if ((got_len = snd_pcm_writei(alsa_handler, data, (len/ao_bps))) != (len/ao_bps)) {     
+    //if ((got_len = snd_pcm_writei(alsa_handler, data, (len/ao_data.bps))) != (len/ao_data.bps)) {     
     //SHOULD BE FIXED      
 	if (got_len == -EPIPE) /* underrun? */
 	{
@@ -498,7 +490,7 @@ static int play(void* data, int len, int flags)
 		printf("alsa-play: playback prepare error: %s\n", snd_strerror(got_len));
 		return(0);
 	    }
-	    if ((got_len = snd_pcm_writei(alsa_handler, data, (len/ao_bps))) != (len/ao_bps))
+	    if ((got_len = snd_pcm_writei(alsa_handler, data, (len/ao_data.bps))) != (len/ao_data.bps))
 	    {
 		printf("alsa-play: write error after reset: %s - giving up\n",
 		    snd_strerror(got_len));
@@ -533,7 +525,7 @@ static int get_space()
 	case SND_PCM_STATE_OPEN:
 	case SND_PCM_STATE_PREPARED:
 	case SND_PCM_STATE_RUNNING:
-	    ret = snd_pcm_status_get_avail(status) * ao_bps;
+	    ret = snd_pcm_status_get_avail(status) * ao_data.bps;
 	    break;
 	default:
 	    ret = 0;
@@ -546,11 +538,11 @@ static int get_space()
     return(ret);
 }
 
-/* how many unplayed bytes are in the buffer */
-static int get_delay()
+/* delay in seconds between first and last sample in buffer */
+static float get_delay()
 {
     snd_pcm_status_t *status;
-    int ret;
+    float ret;
     
     if ((ret = snd_pcm_status_malloc(&status)) < 0)
     {
@@ -569,7 +561,7 @@ static int get_delay()
 	case SND_PCM_STATE_OPEN:
 	case SND_PCM_STATE_PREPARED:
 	case SND_PCM_STATE_RUNNING:
-	    ret = snd_pcm_status_get_delay(status) * ao_bps;
+	    ret = (float)snd_pcm_status_get_delay(status)/(float)ao_data.samplerate;
 	    break;
 	default:
 	    ret = 0;
diff --git a/libao2/ao_mpegpes.c b/libao2/ao_mpegpes.c
index a26811a018..598db3673b 100644
--- a/libao2/ao_mpegpes.c
+++ b/libao2/ao_mpegpes.c
@@ -16,13 +16,6 @@ static ao_info_t info =
 
 LIBAO_EXTERN(mpegpes)
 
-// there are some globals:
-// ao_samplerate
-// ao_channels
-// ao_format
-// ao_bps
-// ao_outburst
-// ao_buffersize
 
 // to set/get/query special features/parameters
 static int control(int cmd,int arg){
@@ -33,8 +26,8 @@ static int control(int cmd,int arg){
 // return: 1=success 0=fail
 static int init(int rate,int channels,int format,int flags){
 
-    ao_outburst=2000;
-    ao_format=format;
+    ao_data.outburst=2000;
+    ao_data.format=format;
 
     return 1;
 }
@@ -67,10 +60,10 @@ extern int vo_pts;
 
 // return: how many bytes can be played without blocking
 static int get_space(){
-    float x=(float)(vo_pts-ao_pts)/90000.0-0.5;
+    float x=(float)(vo_pts-ao_data.pts)/90000.0-0.5;
     int y;
     if(x<=0) return 0;
-    y=48000*4*x;y/=ao_outburst;y*=ao_outburst;
+    y=48000*4*x;y/=ao_data.outburst;y*=ao_data.outburst;
 //    printf("diff: %5.3f -> %d  \n",x,y);
     return y;
 }
@@ -79,20 +72,20 @@ static int get_space(){
 // it should round it down to outburst*n
 // return: number of bytes played
 static int play(void* data,int len,int flags){
-    if(ao_format==AFMT_MPEG)
-	send_pes_packet(data,len,0x1C0,ao_pts);
+    if(ao_data.format==AFMT_MPEG)
+	send_pes_packet(data,len,0x1C0,ao_data.pts);
     else {
 	int i;
 	unsigned short *s=data;
 	for(i=0;i<len/2;i++) s[i]=(s[i]>>8)|(s[i]<<8); // le<->be
-	send_lpcm_packet(data,len,0xA0,ao_pts);
+	send_lpcm_packet(data,len,0xA0,ao_data.pts);
     }
     return len;
 }
 
-// return: how many unplayed bytes are in the buffer
-static int get_delay(){
+// return: delay in seconds between first and last sample in buffer
+static float get_delay(){
 
-    return 0;
+    return 0.0;
 }
 
diff --git a/libao2/ao_null.c b/libao2/ao_null.c
index 8954278f90..e894c47bfb 100644
--- a/libao2/ao_null.c
+++ b/libao2/ao_null.c
@@ -14,13 +14,6 @@ static ao_info_t info =
 
 LIBAO_EXTERN(null)
 
-// there are some globals:
-// ao_samplerate
-// ao_channels
-// ao_format
-// ao_bps
-// ao_outburst
-// ao_buffersize
 
 // to set/get/query special features/parameters
 static int control(int cmd,int arg){
@@ -31,7 +24,7 @@ static int control(int cmd,int arg){
 // return: 1=success 0=fail
 static int init(int rate,int channels,int format,int flags){
 
-    ao_outburst=4096;
+    ao_data.outburst=4096;
 
     return 0;
 }
@@ -61,7 +54,7 @@ static void audio_resume()
 // return: how many bytes can be played without blocking
 static int get_space(){
 
-    return ao_outburst;
+    return ao_data.outburst;
 }
 
 // plays 'len' bytes of 'data'
@@ -72,10 +65,10 @@ static int play(void* data,int len,int flags){
     return len;
 }
 
-// return: how many unplayed bytes are in the buffer
-static int get_delay(){
+// return: delay in seconds between first and last sample in buffer
+static float get_delay(){
 
-    return 0;
+    return 0.0;
 }
 
 
diff --git a/libao2/ao_oss.c b/libao2/ao_oss.c
index b65b1ce679..d6e9c45ac4 100644
--- a/libao2/ao_oss.c
+++ b/libao2/ao_oss.c
@@ -28,14 +28,6 @@ static ao_info_t info =
 
 LIBAO_EXTERN(oss)
 
-// there are some globals:
-// ao_samplerate
-// ao_channels
-// ao_format
-// ao_bps
-// ao_outburst
-// ao_buffersize
-
 static char *dsp="/dev/dsp";
 static audio_buf_info zz;
 static int audio_fd=-1;
@@ -57,7 +49,7 @@ static int control(int cmd,int arg){
 	    ao_control_vol_t *vol = (ao_control_vol_t *)arg;
 	    int fd, v, mcmd, devs;
 
-	    if(ao_format == AFMT_AC3)
+	    if(ao_data.format == AFMT_AC3)
 		return CONTROL_TRUE;
     
 	    if ((fd = open("/dev/mixer", O_RDONLY)) > 0)
@@ -118,58 +110,62 @@ static int init(int rate,int channels,int format,int flags){
     return 0;
   }
 
-  ao_format=format;
-  ioctl (audio_fd, SNDCTL_DSP_SETFMT, &ao_format);
-  if(format == AFMT_AC3 && ao_format != AFMT_AC3) {
+  ao_data.bps=(channels+1)*rate;
+  if(format != AFMT_U8 && format != AFMT_S8)
+    ao_data.bps*=2;
+
+  ao_data.format=format;
+  ioctl (audio_fd, SNDCTL_DSP_SETFMT, &ao_data.format);
+  if(format == AFMT_AC3 && ao_data.format != AFMT_AC3) {
       printf("Can't set audio device %s to AC3 output\n", dsp);
       return 0;
   }
   printf("audio_setup: sample format: %s (requested: %s)\n",
-    audio_out_format_name(ao_format), audio_out_format_name(format));
+    audio_out_format_name(ao_data.format), audio_out_format_name(format));
   
   if(format != AFMT_AC3) {
-  ao_channels=channels-1;
-  ioctl (audio_fd, SNDCTL_DSP_STEREO, &ao_channels);
+  ao_data.channels=channels-1;
+  ioctl (audio_fd, SNDCTL_DSP_STEREO, &ao_data.channels);
   
   // set rate
-  ao_samplerate=rate;
-  ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_samplerate);
-  printf("audio_setup: using %d Hz samplerate (requested: %d)\n",ao_samplerate,rate);
+  ao_data.samplerate=rate;
+  ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
+  printf("audio_setup: using %d Hz samplerate (requested: %d)\n",ao_data.samplerate,rate);
   }
 
   if(ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)==-1){
       int r=0;
       printf("audio_setup: driver doesn't support SNDCTL_DSP_GETOSPACE :-(\n");
       if(ioctl(audio_fd, SNDCTL_DSP_GETBLKSIZE, &r)==-1){
-          printf("audio_setup: %d bytes/frag (config.h)\n",ao_outburst);
+          printf("audio_setup: %d bytes/frag (config.h)\n",ao_data.outburst);
       } else {
-          ao_outburst=r;
-          printf("audio_setup: %d bytes/frag (GETBLKSIZE)\n",ao_outburst);
+          ao_data.outburst=r;
+          printf("audio_setup: %d bytes/frag (GETBLKSIZE)\n",ao_data.outburst);
       }
   } else {
       printf("audio_setup: frags: %3d/%d  (%d bytes/frag)  free: %6d\n",
           zz.fragments, zz.fragstotal, zz.fragsize, zz.bytes);
-      if(ao_buffersize==-1) ao_buffersize=zz.bytes;
-      ao_outburst=zz.fragsize;
+      if(ao_data.buffersize==-1) ao_data.buffersize=zz.bytes;
+      ao_data.outburst=zz.fragsize;
   }
 
-  if(ao_buffersize==-1){
+  if(ao_data.buffersize==-1){
     // Measuring buffer size:
     void* data;
-    ao_buffersize=0;
+    ao_data.buffersize=0;
 #ifdef HAVE_AUDIO_SELECT
-    data=malloc(ao_outburst); memset(data,0,ao_outburst);
-    while(ao_buffersize<0x40000){
+    data=malloc(ao_data.outburst); memset(data,0,ao_data.outburst);
+    while(ao_data.buffersize<0x40000){
       fd_set rfds;
       struct timeval tv;
       FD_ZERO(&rfds); FD_SET(audio_fd,&rfds);
       tv.tv_sec=0; tv.tv_usec = 0;
       if(!select(audio_fd+1, NULL, &rfds, NULL, &tv)) break;
-      write(audio_fd,data,ao_outburst);
-      ao_buffersize+=ao_outburst;
+      write(audio_fd,data,ao_data.outburst);
+      ao_data.buffersize+=ao_data.outburst;
     }
     free(data);
-    if(ao_buffersize==0){
+    if(ao_data.buffersize==0){
         printf("\n   ***  Your audio driver DOES NOT support select()  ***\n");
           printf("Recompile mplayer with #undef HAVE_AUDIO_SELECT in config.h !\n\n");
         return 0;
@@ -197,10 +193,10 @@ static void reset(){
 	return;
     }
 
-  ioctl (audio_fd, SNDCTL_DSP_SETFMT, &ao_format);
-  if(ao_format != AFMT_AC3) {
-  ioctl (audio_fd, SNDCTL_DSP_STEREO, &ao_channels);
-  ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_samplerate);
+  ioctl (audio_fd, SNDCTL_DSP_SETFMT, &ao_data.format);
+  if(ao_data.format != AFMT_AC3) {
+  ioctl (audio_fd, SNDCTL_DSP_STEREO, &ao_data.channels);
+  ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
   }
 }
 
@@ -219,7 +215,7 @@ static void audio_resume()
 
 // return: how many bytes can be played without blocking
 static int get_space(){
-  int playsize=ao_outburst;
+  int playsize=ao_data.outburst;
 
 #ifdef SNDCTL_DSP_GETOSPACE
   if(ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)!=-1){
@@ -240,35 +236,38 @@ static int get_space(){
     }
 #endif
 
-  return ao_outburst;
+  return ao_data.outburst;
 }
 
 // plays 'len' bytes of 'data'
 // it should round it down to outburst*n
 // return: number of bytes played
 static int play(void* data,int len,int flags){
-    len/=ao_outburst;
-    len=write(audio_fd,data,len*ao_outburst);
+    len/=ao_data.outburst;
+    len=write(audio_fd,data,len*ao_data.outburst);
     return len;
 }
 
 static int audio_delay_method=2;
 
-// return: how many unplayed bytes are in the buffer
-static int get_delay(){
+// return: delay in seconds between first and last sample in buffer
+static float get_delay(){
+  /* Calculate how many bytes/second is sent out */
   if(audio_delay_method==2){
-      // 
       int r=0;
       if(ioctl(audio_fd, SNDCTL_DSP_GETODELAY, &r)!=-1)
-         return r;
+         return ((float)r)/(float)ao_data.bps;
       audio_delay_method=1; // fallback if not supported
   }
   if(audio_delay_method==1){
       // SNDCTL_DSP_GETOSPACE
       if(ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)!=-1)
-         return ao_buffersize-zz.bytes;
+         return ((float)(ao_data.buffersize-zz.bytes))/(float)ao_data.bps;
       audio_delay_method=0; // fallback if not supported
   }
-  return ao_buffersize;
+  return ((float)ao_data.buffersize)/(float)ao_data.bps;
 }
 
+
+
+
diff --git a/libao2/ao_pcm.c b/libao2/ao_pcm.c
index 6cc74873e4..6f6179c5fc 100644
--- a/libao2/ao_pcm.c
+++ b/libao2/ao_pcm.c
@@ -14,14 +14,6 @@ static ao_info_t info =
 
 LIBAO_EXTERN(pcm)
 
-// there are some globals:
-// ao_samplerate
-// ao_channels
-// ao_format
-// ao_bps
-// ao_outburst
-// ao_buffersize
-
 char *ao_outputfilename = NULL;
 int ao_pcm_waveheader = 1;
 
@@ -90,7 +82,7 @@ static int init(int rate,int channels,int format,int flags){
 	printf("PCM: Info - to write WAVE files use -waveheader (default), for RAW PCM -nowaveheader.\n");
 	fp = fopen(ao_outputfilename, "wb");
 
-	ao_outburst = 4096;
+	ao_data.outburst = 4096;
 
 
 	if(fp) {
@@ -134,7 +126,7 @@ static void audio_resume()
 // return: how many bytes can be played without blocking
 static int get_space(){
 
-    return ao_outburst;
+    return ao_data.outburst;
 }
 
 // plays 'len' bytes of 'data'
@@ -151,10 +143,10 @@ static int play(void* data,int len,int flags){
 	return len;
 }
 
-// return: how many unplayed bytes are in the buffer
-static int get_delay(){
+// return: delay in seconds between first and last sample in buffer
+static float get_delay(){
 
-    return 0;
+    return 0.0;
 }
 
 
diff --git a/libao2/ao_sdl.c b/libao2/ao_sdl.c
index 2d6a430aec..616761a95c 100644
--- a/libao2/ao_sdl.c
+++ b/libao2/ao_sdl.c
@@ -29,13 +29,6 @@ static ao_info_t info =
 
 LIBAO_EXTERN(sdl)
 
-// there are some globals:
-// ao_samplerate
-// ao_channels
-// ao_format
-// ao_bps
-// ao_outburst
-// ao_buffersize
 
 extern int verbose;
 
@@ -150,6 +143,10 @@ static int init(int rate,int channels,int format,int flags){
 		setenv("SDL_AUDIODRIVER", ao_subdevice, 1);
 		printf("SDL: using %s audio driver\n", ao_subdevice);
 	}
+
+	ao_data.bps=(channels+1)*rate;
+	if(format != AFMT_U8 && format != AFMT_S8)
+	  ao_data.bps*=2;
 	
 	/* The desired audio format (see SDL_AudioSpec) */
 	switch(format) {
@@ -205,7 +202,7 @@ void callback(void *userdata, Uint8 *stream, int len); userdata is the pointer s
 	} 
 	
 	if(verbose) printf("SDL: buf size = %d\n",aspec.size);
-	if(ao_buffersize==-1) ao_buffersize=aspec.size;
+	if(ao_data.buffersize==-1) ao_data.buffersize=aspec.size;
 	
 	/* unsilence audio, if callback is ready */
 	SDL_PauseAudio(0);
@@ -278,9 +275,9 @@ static int play(void* data,int len,int flags){
 #endif
 }
 
-// return: how many unplayed bytes are in the buffer
-static int get_delay(){
-    return buffered_bytes + ao_buffersize;
+// return: delay in seconds between first and last sample in buffer
+static float get_delay(){
+    return (float)(buffered_bytes + ao_data.buffersize)/(float)ao_data.bps;
 }
 
 
diff --git a/libao2/ao_sgi.c b/libao2/ao_sgi.c
index 74430955af..5dfc6124ed 100644
--- a/libao2/ao_sgi.c
+++ b/libao2/ao_sgi.c
@@ -22,13 +22,6 @@ static ao_info_t info =
 
 LIBAO_EXTERN(sgi)
 
-// there are some globals:
-// ao_samplerate
-// ao_channels
-// ao_format
-// ao_bps
-// ao_outburst
-// ao_buffersize
 
 static ALconfig	ao_config;
 static ALport	ao_port;
@@ -86,9 +79,9 @@ static int init(int rate, int channels, int format, int flags) {
 
   }
   
-  ao_buffersize=131072;
-  ao_outburst = ao_buffersize/16;
-  ao_channels = channels;
+  ao_data.buffersize=131072;
+  ao_data.outburst = ao_data.buffersize/16;
+  ao_data.channels = channels;
   
   ao_config = alNewConfig();
   
@@ -164,7 +157,7 @@ static int get_space() {
   // printf("ao_sgi, get_space: (ao_outburst %d)\n", ao_outburst);
   // printf("ao_sgi, get_space: alGetFillable [%d] \n", alGetFillable(ao_port));
   
-  return alGetFillable(ao_port)*(2*ao_channels);
+  return alGetFillable(ao_port)*(2*ao_data.channels);
     
 }
 
@@ -177,14 +170,14 @@ static int play(void* data, int len, int flags) {
   // printf("ao_sgi, play: len %d flags %d (%d %d)\n", len, flags, ao_port, ao_config);
   // printf("channels %d\n", ao_channels);
 
-  alWriteFrames(ao_port, data, len/(2*ao_channels));
+  alWriteFrames(ao_port, data, len/(2*ao_data.channels));
   
   return len;
   
 }
 
-// return: how many unplayed bytes are in the buffer
-static int get_delay(){
+// return: delay in seconds between first and last sample in buffer
+static float get_delay(){
   
   // printf("ao_sgi, get_delay: (ao_buffersize %d)\n", ao_buffersize);