diff --git a/src/audio.c b/src/audio.c
index 3dfa76ab..c1473dde 100644
--- a/src/audio.c
+++ b/src/audio.c
@@ -329,12 +329,16 @@ void flush_output(int outmode, audio_output_t *ao, unsigned char *bytes, size_t
 	}
 }
 
+/* is this used? */
 void close_output(int outmode, audio_output_t *ao)
 {
 
     switch(outmode) {
       case DECODE_AUDIO:
         ao->close(ao);
+        /* Module frees and closes its resources, but may not reset them. */
+        ao->userptr = NULL;
+        ao->fn = -1;
         break;
       case DECODE_WAV:
         wav_close();
@@ -348,3 +352,14 @@ void close_output(int outmode, audio_output_t *ao)
     }
 
 }
+
+/* Also for WAV decoding? */
+int reset_output(audio_output_t *ao)
+{
+	if(param.outmode == DECODE_AUDIO)
+	{
+		close_output(param.outmode, ao);
+		return ao->open(ao);
+	}
+	else return 0;
+}
diff --git a/src/audio.h b/src/audio.h
index 9c7b4cfa..d7dba44a 100644
--- a/src/audio.h
+++ b/src/audio.h
@@ -78,7 +78,7 @@ const char* audio_encoding_name(const int encoding, const int longer);
 int init_output(audio_output_t *ao, mpg123_handle *mh);
 void flush_output(int outmode, audio_output_t *ao, unsigned char *bytes, size_t count);
 void close_output(int mod, audio_output_t *ao );
-
+int reset_output(audio_output_t *ao);
 
 #endif
 
diff --git a/src/buffer.c b/src/buffer.c
index fafaf66c..c4a48531 100644
--- a/src/buffer.c
+++ b/src/buffer.c
@@ -141,15 +141,12 @@ void buffer_loop(audio_output_t *ao, sigset_t *oldsigset)
 			 */
 			if (xf->wakeme[XF_WRITER])
 				xfermem_putcmd(my_fd, XF_CMD_WAKEUP);
-			if (param.outmode == DECODE_AUDIO) {
-				ao->close(ao);
-				ao->rate = xf->buf[0]; 
-				ao->channels = xf->buf[1]; 
-				ao->format = xf->buf[2];
-				if (ao->open(ao) < 0) {
-					perror("audio");
-					exit(1);
-				}
+			ao->rate = xf->buf[0]; 
+			ao->channels = xf->buf[1]; 
+			ao->format = xf->buf[2];
+			if (reset_output(ao) < 0) {
+				error1("failed to reset audio: %s", strerror(errno));
+				exit(1);
 			}
 		}
 		if ( (bytes = xfermem_get_usedspace(xf)) < outburst ) {
diff --git a/src/mpg123.c b/src/mpg123.c
index af0c997b..1fa4791a 100644
--- a/src/mpg123.c
+++ b/src/mpg123.c
@@ -419,9 +419,9 @@ static void reset_audio(void)
 		 *   the device's internal buffer before
 		 *   changing the sample rate.   [OF]
 		 */
-		ao->close(ao);
-		if (ao->open(ao) < 0) {
-			error("failed to open audio device");
+		if(reset_output(ao) < 0)
+		{
+			error1("failed to reset audio device: %s", strerror(errno));
 			safe_exit(1);
 		}
 	}
diff --git a/src/output/Makefile.am b/src/output/Makefile.am
index c5156629..cf03573f 100644
--- a/src/output/Makefile.am
+++ b/src/output/Makefile.am
@@ -21,6 +21,7 @@ output_dummy_la_LDFLAGS = -module -avoid-version
 #pkglib_LTLIBRARIES += output_alib.la
 #output_alib_la_SOURCES = alib.c
 #output_alib_la_LDFLAGS = -module -avoid-version
+#output_alib_la_CFLAGS= -I/opt/audio/include
 #endif
 
 if HAVE_ALSA
@@ -90,7 +91,7 @@ if HAVE_PORTAUDIO
 pkglib_LTLIBRARIES += output_portaudio.la
 output_portaudio_la_SOURCES = portaudio.c
 output_portaudio_la_LDFLAGS = -module -avoid-version
-output_portaudio_la_LIBADD  = -lportaudio
+output_portaudio_la_LIBADD  = -lportaudio @PORTAUDIO_EXTRA_LIB@
 endif
 
 if HAVE_PULSE
@@ -109,11 +110,12 @@ output_sdl_la_LDFLAGS = -module -avoid-version @SDL_LDFLAGS@
 output_sdl_la_LIBADD = @SDL_LIBS@
 endif
 
-#if HAVE_SGI
-#pkglib_LTLIBRARIES += output_sgi.la
-#output_sgi_la_SOURCES = sgi.c
-#output_sgi_la_LDFLAGS = -module -avoid-version
-#endif
+if HAVE_SGI
+pkglib_LTLIBRARIES += output_sgi.la
+output_sgi_la_SOURCES = sgi.c
+output_sgi_la_LDFLAGS = -module -avoid-version
+output_sgi_la_LIBADD = -laudio
+endif
 
 if HAVE_SUN
 pkglib_LTLIBRARIES += output_sun.la
@@ -121,8 +123,9 @@ output_sun_la_SOURCES = sun.c
 output_sun_la_LDFLAGS = -module -avoid-version
 endif
 
-#if HAVE_WIN32
-#pkglib_LTLIBRARIES += output_win32.la
-#output_win32_la_SOURCES = win32.c
-#output_win32_la_LDFLAGS = -module -avoid-version
-#endif
+if HAVE_WIN32
+pkglib_LTLIBRARIES += output_win32.la
+output_win32_la_SOURCES = win32.c
+output_win32_la_LDFLAGS = -module -avoid-version
+output_win32_la_LIBADD= -lwinmm
+endif
diff --git a/src/output/alsa.c b/src/output/alsa.c
index b679d6fe..032efb98 100644
--- a/src/output/alsa.c
+++ b/src/output/alsa.c
@@ -137,6 +137,7 @@ static int open_alsa(audio_output_t *ao)
 {
 	const char *pcm_name;
 	snd_pcm_t *pcm=NULL;
+	debug1("open_alsa with %p", ao->userptr);
 
 	pcm_name = ao->device ? ao->device : "default";
 	if (snd_pcm_open(&pcm, pcm_name, SND_PCM_STREAM_PLAYBACK, 0) < 0) {
@@ -215,11 +216,12 @@ static void flush_alsa(audio_output_t *ao)
 static int close_alsa(audio_output_t *ao)
 {
 	snd_pcm_t *pcm=(snd_pcm_t*)ao->userptr;
-
+	debug1("close_alsa with %p", ao->userptr);
 	if(pcm != NULL) /* be really generous for being called without any device opening */
 	{
 		if (snd_pcm_state(pcm) == SND_PCM_STATE_RUNNING)
 			snd_pcm_drain(pcm);
+		ao->userptr = NULL; /* Should alsa do this or the module wrapper? */
 		return snd_pcm_close(pcm);
 	}
 	else return 0;