rtsp-media: Unblock all streams

When unsuspending and going to PLAYING, unblock all streams instead of
only those that are linked (the linked streams are the ones for which
SETUP has been called). GST_FLOW_NOT_LINKED will be returned when
pushing buffers on unlinked streams.

This change is because playback using single-threaded demuxers like
matroska-demux could be blocked if SETUP was not called for all media.
Demuxers that use GstFlowCombiner (including gstoggdemux, gstavidemux,
gstflvdemux, qtdemux, and matroska-demux) will handle
GST_FLOW_NOT_LINKED automatically.

Fixes #39

diff --git a/gst/rtsp-server/rtsp-media.c b/gst/rtsp-server/rtsp-media.c
index 45463f6..345559d 100644 (file)
--- a/gst/rtsp-server/rtsp-media.c
+++ b/gst/rtsp-server/rtsp-media.c
@@ -2635,15 +2635,15 @@ media_streams_set_blocked (GstRTSPMedia * media, gboolean blocked)
 static void
 stream_unblock (GstRTSPStream * stream, GstRTSPMedia * media)
 {
-  gst_rtsp_stream_unblock_linked (stream);
+  gst_rtsp_stream_set_blocked (stream, FALSE);
 }
 
 static void
-media_unblock_linked (GstRTSPMedia * media)
+media_unblock (GstRTSPMedia * media)
 {
   GstRTSPMediaPrivate *priv = media->priv;
 
-  GST_DEBUG ("media %p unblocking linked streams", media);
+  GST_DEBUG ("media %p unblocking streams", media);
   /* media is not blocked any longer, as it contains active streams,
    * streams that are complete */
   priv->blocked = FALSE;
@@ -4393,8 +4393,8 @@ default_unsuspend (GstRTSPMedia * media)
         gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
         /* at this point the media pipeline has been updated and contain all
          * specific transport parts: all active streams contain at least one sink
-         * element and it's safe to unblock any blocked streams that are active */
-        media_unblock_linked (media);
+         * element and it's safe to unblock all blocked streams */
+        media_unblock (media);
       } else {
         /* streams are not blocked and media is suspended from PAUSED */
         gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
@@ -4414,8 +4414,8 @@ default_unsuspend (GstRTSPMedia * media)
       gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
       /* at this point the media pipeline has been updated and contain all
        * specific transport parts: all active streams contain at least one sink
-       * element and it's safe to unblock any blocked streams that are active */
-      media_unblock_linked (media);
+       * element and it's safe to unblock all blocked streams */
+      media_unblock (media);
       if (!start_preroll (media))
         goto start_failed;
 
@@ -4507,7 +4507,7 @@ media_set_pipeline_state_locked (GstRTSPMedia * media, GstState state)
     } else {
       if (state == GST_STATE_PLAYING)
         /* make sure pads are not blocking anymore when going to PLAYING */
-        media_unblock_linked (media);
+        media_unblock (media);
 
       if (state == GST_STATE_PAUSED) {
         set_state_ret = set_state (media, state);

diff --git a/tests/check/gst/rtspserver.c b/tests/check/gst/rtspserver.c
index da6df2d..0a0eec4 100644 (file)
--- a/tests/check/gst/rtspserver.c
+++ b/tests/check/gst/rtspserver.c
@@ -30,10 +30,14 @@
 
 #include "rtsp-server.h"
 
+#define ERRORIGNORE "errorignore ignore-error=false ignore-notlinked=true " \
+  "ignore-notnegotiated=false convert-to=ok"
 #define VIDEO_PIPELINE "videotestsrc ! " \
+  ERRORIGNORE " ! " \
   "video/x-raw,width=352,height=288 ! " \
   "rtpgstpay name=pay0 pt=96"
 #define AUDIO_PIPELINE "audiotestsrc ! " \
+  ERRORIGNORE " ! " \
   "audio/x-raw,rate=8000 ! " \
   "rtpgstpay name=pay1 pt=97"