Fixed some plugins

Original commit message from CVS:
Fixed some plugins

7 files changed, 21 insertions, 31 deletions

diff --git a/ext/audiofile/gstafsrc.c b/ext/audiofile/gstafsrc.c
index b6538a6d..0d4e261f 100644
--- a/ext/audiofile/gstafsrc.c
+++ b/ext/audiofile/gstafsrc.c
@@ -215,8 +215,8 @@ gst_afsrc_get (GstPad *pad)
  	                    frameCount);
   readbytes = readframes * (src->channels * src->width / 8);
   if (readbytes == 0) {
-    gst_element_signal_eos (GST_ELEMENT (src));
-      return NULL;  
+    gst_element_set_eos (GST_ELEMENT (src));
+    return GST_BUFFER (gst_event_new (GST_EVENT_EOS));  
   }
   
   GST_BUFFER_SIZE (buf) = readbytes;
diff --git a/ext/sdl/sdlvideosink.c b/ext/sdl/sdlvideosink.c
index 10e1ca7d..bbc2f322 100644
--- a/ext/sdl/sdlvideosink.c
+++ b/ext/sdl/sdlvideosink.c
@@ -239,19 +239,17 @@ gst_sdlvideosink_chain (GstPad *pad, GstBuffer *buf)
 
   sdlvideosink = GST_SDLVIDEOSINK (gst_pad_get_parent (pad));
 
-  if (!GST_BUFFER_FLAG_IS_SET(buf, GST_BUFFER_FLUSH)) {
-    GST_DEBUG (0,"videosink: clock wait: %llu\n", GST_BUFFER_TIMESTAMP(buf));
+  GST_DEBUG (0,"videosink: clock wait: %llu\n", GST_BUFFER_TIMESTAMP(buf));
 
-    jitter = gst_clock_current_diff(sdlvideosink->clock, GST_BUFFER_TIMESTAMP (buf));
+  jitter = gst_clock_current_diff(sdlvideosink->clock, GST_BUFFER_TIMESTAMP (buf));
 
-    if (jitter > 500000 || jitter < -500000)
-    {
-      GST_DEBUG (0, "jitter: %lld\n", jitter);
-      gst_clock_set (sdlvideosink->clock, GST_BUFFER_TIMESTAMP (buf));
-    }
-    else {
-      gst_clock_wait(sdlvideosink->clock, GST_BUFFER_TIMESTAMP(buf), GST_OBJECT(sdlvideosink));
-    }
+  if (jitter > 500000 || jitter < -500000)
+  {
+    GST_DEBUG (0, "jitter: %lld\n", jitter);
+    gst_clock_set (sdlvideosink->clock, GST_BUFFER_TIMESTAMP (buf));
+  }
+  else {
+    gst_clock_wait(sdlvideosink->clock, GST_BUFFER_TIMESTAMP(buf), GST_OBJECT(sdlvideosink));
   }
 
   /* Lock SDL/yuv-overlay */
diff --git a/gst/flx/gstflxdec.c b/gst/flx/gstflxdec.c
index 211ab476..5fd5eb23 100644
--- a/gst/flx/gstflxdec.c
+++ b/gst/flx/gstflxdec.c
@@ -571,7 +571,6 @@ gst_flxdec_loop (GstElement *element)
              GST_BUFFER_DATA(flxdec->frame),
              GST_BUFFER_DATA(flxdec->out));
 
-        //GST_BUFFER_FLAG_SET(flxdec->out, GST_BUFFER_FLUSH);
         gst_pad_push(flxdec->srcpad, flxdec->out);
         
         break;
diff --git a/gst/mpeg1videoparse/gstmp1videoparse.c b/gst/mpeg1videoparse/gstmp1videoparse.c
index 74d910e4..39fdfa29 100644
--- a/gst/mpeg1videoparse/gstmp1videoparse.c
+++ b/gst/mpeg1videoparse/gstmp1videoparse.c
@@ -252,10 +252,11 @@ gst_mp1videoparse_real_chain (Mp1VideoParse *mp1videoparse, GstBuffer *buf, GstP
 
   time_stamp = GST_BUFFER_TIMESTAMP(buf);
 
-
-  if (GST_BUFFER_FLAG_IS_SET(buf, GST_BUFFER_FLUSH)) {
+  /* FIXME, handle events here */
+  /*
     gst_mp1videoparse_flush(mp1videoparse);
-  }
+    */
+ 
 
   if (mp1videoparse->partialbuf) {
     offset = GST_BUFFER_SIZE(mp1videoparse->partialbuf);
@@ -349,7 +350,7 @@ gst_mp1videoparse_real_chain (Mp1VideoParse *mp1videoparse, GstBuffer *buf, GstP
     GST_BUFFER_TIMESTAMP(outbuf) = mp1videoparse->last_pts;
 
     if (mp1videoparse->in_flush) {
-      GST_BUFFER_FLAG_SET(outbuf, GST_BUFFER_FLUSH);
+      /* FIXME, send a flush event here */
       mp1videoparse->in_flush = FALSE;
     }
 
diff --git a/gst/mpegaudioparse/gstmpegaudioparse.c b/gst/mpegaudioparse/gstmpegaudioparse.c
index f8ece389..220d6dd2 100644
--- a/gst/mpegaudioparse/gstmpegaudioparse.c
+++ b/gst/mpegaudioparse/gstmpegaudioparse.c
@@ -247,13 +247,14 @@ gst_mp3parse_chain (GstPad *pad, GstBuffer *buf)
 
   last_ts = GST_BUFFER_TIMESTAMP(buf);
 
-  if (GST_BUFFER_FLAG_IS_SET(buf, GST_BUFFER_FLUSH)) {
+  /* FIXME, do flush */
+  /*
     if (mp3parse->partialbuf) {
       gst_buffer_unref(mp3parse->partialbuf);
       mp3parse->partialbuf = NULL;
     }
     mp3parse->in_flush = TRUE;
-  }
+    */
 
   // if we have something left from the previous frame
   if (mp3parse->partialbuf) {
@@ -329,12 +330,9 @@ gst_mp3parse_chain (GstPad *pad, GstBuffer *buf)
 	if (mp3parse->skip == 0) {
           GST_DEBUG (0,"mp3parse: pushing buffer of %d bytes\n",GST_BUFFER_SIZE(outbuf));
 	  if (mp3parse->in_flush) {
-	    GST_BUFFER_FLAG_SET(outbuf, GST_BUFFER_FLUSH);
+	    /* FIXME do some sort of flush event */
 	    mp3parse->in_flush = FALSE;
 	  }
-	  else {
-	    GST_BUFFER_FLAG_UNSET(outbuf, GST_BUFFER_FLUSH);
-	  }
 	  GST_BUFFER_TIMESTAMP(outbuf) = last_ts;
           gst_pad_push(mp3parse->srcpad,outbuf);
 	}
diff --git a/gst/speed/filter.func b/gst/speed/filter.func
index 8d8a911b..cf23725b 100644
--- a/gst/speed/filter.func
+++ b/gst/speed/filter.func
@@ -39,7 +39,7 @@
           while (GST_IS_EVENT(in)) {
             switch (GST_EVENT_TYPE(in)) {
             case GST_EVENT_EOS:
-              gst_element_set_state((GstElement*)filter, GST_STATE_PAUSED);
+              gst_element_set_eos((GstElement*)filter);
               gst_pad_push(filter->srcpad, in);
               return;
             default:
diff --git a/sys/vcd/vcdsrc.c b/sys/vcd/vcdsrc.c
index e6fb2e90..47305878 100644
--- a/sys/vcd/vcdsrc.c
+++ b/sys/vcd/vcdsrc.c
@@ -262,12 +262,6 @@ vcdsrc_get (GstPad *pad)
     goto read_sector;
   }
 
-/*  if (readbytes == 0) { */
-/*    gst_src_signal_eos(GST_SRC(vcdsrc)); */
-/*    gst_element_set_state(GST_ELEMENT(vcdsrc),~GST_STATE_PLAYING); */
-/*    gst_element_set_state(GST_ELEMENT(vcdsrc),~GST_STATE_RUNNING); */
-/*    return;							     */
-/*  }								     */
 
   GST_BUFFER_OFFSET(buf) = vcdsrc->curoffset;
   GST_BUFFER_SIZE(buf) = vcdsrc->bytes_per_read;