From: Tim-Philipp Müller <tim@centricular.net>
Date: Fri, 9 Feb 2007 16:24:45 +0000 (+0000)
Subject: ext/lame/gstlame.*: On receiving EOS, we try to push a last buffer with the remaining... 
X-Git-Tag: 1.19.3~505^2~1809
X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=0d4e832887cc67d15e8e7c72ea64c47a0a883237;p=platform%2Fupstream%2Fgstreamer.git

ext/lame/gstlame.*: On receiving EOS, we try to push a last buffer with the remaining samples. Don't do that if we go...

Original commit message from CVS:
* ext/lame/gstlame.c: (gst_lame_sink_event), (gst_lame_chain),
(gst_lame_change_state):
* ext/lame/gstlame.h:
On receiving EOS, we try to push a last buffer with the remaining
samples. Don't do that if we got an unclean flow return on the last
gst_pad_push(), downstream might not handle this very gracefully
(see #403168).
* gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_chain):
Pass flow returns upstream (helps #403168).
---

diff --git a/ChangeLog b/ChangeLog
index e2555ac..d7d4f66 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,3 +1,16 @@
+2007-02-09  Tim-Philipp Müller  <tim at centricular dot net>
+
+	* ext/lame/gstlame.c: (gst_lame_sink_event), (gst_lame_chain),
+	(gst_lame_change_state):
+	* ext/lame/gstlame.h:
+	  On receiving EOS, we try to push a last buffer with the remaining
+	  samples. Don't do that if we got an unclean flow return on the last
+	  gst_pad_push(), downstream might not handle this very gracefully
+	  (see #403168).
+
+	* gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_chain):
+	  Pass flow returns upstream (helps #403168).
+
 2007-02-09  Stefan Kost  <ensonic@users.sf.net>
 
 	* gst/synaesthesia/gstsynaesthesia.c:
diff --git a/ext/lame/gstlame.c b/ext/lame/gstlame.c
index 92f8bef..24853da 100644
--- a/ext/lame/gstlame.c
+++ b/ext/lame/gstlame.c
@@ -920,7 +920,7 @@ gst_lame_sink_event (GstPad * pad, GstEvent * event)
         buf = gst_buffer_new_and_alloc (7200);
         size = lame_encode_flush (lame->lgf, GST_BUFFER_DATA (buf), 7200);
 
-        if (size > 0) {
+        if (size > 0 && lame->last_flow == GST_FLOW_OK) {
           gint64 duration;
 
           duration = gst_util_uint64_scale_int (size, GST_SECOND,
@@ -941,7 +941,8 @@ gst_lame_sink_event (GstPad * pad, GstEvent * event)
           gst_buffer_set_caps (buf, GST_PAD_CAPS (lame->srcpad));
           gst_pad_push (lame->srcpad, buf);
         } else {
-          GST_DEBUG_OBJECT (lame, "no final packet (size=%d)", size);
+          GST_DEBUG_OBJECT (lame, "no final packet (size=%d, last_flow=%s)",
+              size, gst_flow_get_name (lame->last_flow));
           gst_buffer_unref (buf);
         }
       }
@@ -1064,6 +1065,7 @@ gst_lame_chain (GstPad * pad, GstBuffer * buf)
     gst_buffer_set_caps (outbuf, GST_PAD_CAPS (lame->srcpad));
 
     result = gst_pad_push (lame->srcpad, outbuf);
+    lame->last_flow = result;
     if (result != GST_FLOW_OK) {
       GST_DEBUG_OBJECT (lame, "flow return: %s", gst_flow_get_name (result));
     }
@@ -1201,6 +1203,7 @@ gst_lame_change_state (GstElement * element, GstStateChange transition)
 
   switch (transition) {
     case GST_STATE_CHANGE_READY_TO_PAUSED:
+      lame->last_flow = GST_FLOW_OK;
       lame->last_ts = GST_CLOCK_TIME_NONE;
       lame->eos_ts = GST_CLOCK_TIME_NONE;
       break;
diff --git a/ext/lame/gstlame.h b/ext/lame/gstlame.h
index 71bfacc..bcae339 100644
--- a/ext/lame/gstlame.h
+++ b/ext/lame/gstlame.h
@@ -93,6 +93,9 @@ struct _GstLame {
   gboolean emphasis;
   gint preset;
 
+  /* track this so we don't send a last buffer in eos handler after error */
+  GstFlowReturn  last_flow;
+
   lame_global_flags *lgf;
 
   /* time tracker */
diff --git a/gst/mpegaudioparse/gstmpegaudioparse.c b/gst/mpegaudioparse/gstmpegaudioparse.c
index 6e52b57..68b62fd 100644
--- a/gst/mpegaudioparse/gstmpegaudioparse.c
+++ b/gst/mpegaudioparse/gstmpegaudioparse.c
@@ -319,6 +319,7 @@ gst_mp3parse_sink_event (GstPad * pad, GstEvent * event)
 static GstFlowReturn
 gst_mp3parse_chain (GstPad * pad, GstBuffer * buf)
 {
+  GstFlowReturn flow = GST_FLOW_OK;
   GstMPEGAudioParse *mp3parse;
   const guchar *data;
   guint32 header;
@@ -327,10 +328,9 @@ gst_mp3parse_chain (GstPad * pad, GstBuffer * buf)
   GstClockTime timestamp;
   guint available;
 
-  mp3parse = GST_MP3PARSE (gst_pad_get_parent (pad));
+  mp3parse = GST_MP3PARSE (GST_PAD_PARENT (pad));
 
-  GST_DEBUG_OBJECT (mp3parse, "received buffer of %d bytes",
-      GST_BUFFER_SIZE (buf));
+  GST_LOG_OBJECT (mp3parse, "buffer of %d bytes", GST_BUFFER_SIZE (buf));
 
   timestamp = GST_BUFFER_TIMESTAMP (buf);
 
@@ -472,7 +472,7 @@ gst_mp3parse_chain (GstPad * pad, GstBuffer * buf)
 
           gst_buffer_set_caps (outbuf, GST_PAD_CAPS (mp3parse->srcpad));
 
-          gst_pad_push (mp3parse->srcpad, outbuf);
+          flow = gst_pad_push (mp3parse->srcpad, outbuf);
 
         } else {
           GST_DEBUG_OBJECT (mp3parse, "skipping buffer of %d bytes",
@@ -486,11 +486,12 @@ gst_mp3parse_chain (GstPad * pad, GstBuffer * buf)
       gst_adapter_flush (mp3parse->adapter, 1);
       GST_DEBUG_OBJECT (mp3parse, "wrong header, skipping byte");
     }
-  }
 
-  gst_object_unref (mp3parse);
+    if (GST_FLOW_IS_FATAL (flow))
+      break;
+  }
 
-  return GST_FLOW_OK;
+  return flow;
 }
 
 static gboolean