[gst-devel] appsink-src example memory leak

[Wim Taymans]

On Mon, 2009-04-27 at 15:59 +0200, mattias wrote:
> Hi,
> 
> I tested to compile the appsink-src example from libgstapp and it
> seems to leak an enormous amount of memory (In fact it seems not to
> free any of the frames).

It does not leak, your example creates as many frames as possible and
then queues them in appsrc (which has by default an unlimited queue
size).  

What you likely want is to make appsrc block with the 'block' property,
this should rate limit the producer pipeline.

Wim

> 
> How could this be avoided, please? I would like to use this mechanism
> for an application and I think that the concept is very good.
> 
> Thanks a lot,
> 
> Mattias
> 
> Here follows the appsink-src.c example:
> 
> ----
> 
> #include <gst/gst.h>
> 
> #include <string.h>
> 
> #include <gst/app/gstappsrc.h>
> #include <gst/app/gstappsink.h>
> #include <gst/app/gstappbuffer.h>
> 
> /* these are the caps we are going to pass through the appsink and
> appsrc */
> 
> const gchar *video_caps =
>   "video/x-raw-yuv,width=640,height=480,framerate=15/1";
> 
> 
> typedef struct
> {
>   GMainLoop *loop;
>   GstElement *source;
>   GstElement *sink;
> } ProgramData;
> 
> /* called when the appsink notifies us that there is a new buffer
> ready for
>  * processing */
> static void
> on_new_buffer_from_source (GstElement * elt, ProgramData * data)
> {
>   guint size;
>   gpointer raw_buffer;
>   GstBuffer *app_buffer, *buffer;
>   GstElement *source;
> 
>   /* get the buffer from appsink */
>   buffer = gst_app_sink_pull_buffer (GST_APP_SINK (elt));
> 
>   /* turn it into an app buffer, it's not really needed, we could
> simply push
>    * the retrieved buffer from appsink into appsrc just fine.  */
>   size = GST_BUFFER_SIZE (buffer);
>   g_print ("Pushing a buffer of size %d\n", size);
>   raw_buffer = g_malloc0 (size);
>   memcpy (raw_buffer, GST_BUFFER_DATA (buffer), size);
>   app_buffer = gst_app_buffer_new (raw_buffer, size, g_free,
> raw_buffer);
> 
>   /* newer basesrc will set caps for use automatically but it does not
> really
>    * hurt to set it on the buffer again */
>   gst_buffer_set_caps (app_buffer, GST_BUFFER_CAPS (buffer));
> 
>   /* we don't need the appsink buffer anymore */
>   gst_buffer_unref (buffer);
> 
>   /* get source an push new buffer */
>   source = gst_bin_get_by_name (GST_BIN (data->sink), "testsource");
>   gst_app_src_push_buffer (GST_APP_SRC (source), app_buffer);
>  
> }
> 
> /* called when we get a GstMessage from the source pipeline when we
> get EOS, we
>  * notify the appsrc of it. */
> static gboolean
> on_source_message (GstBus * bus, GstMessage * message, ProgramData *
> data)
> {
>   GstElement *source;
> 
>   switch (GST_MESSAGE_TYPE (message)) {
>     case GST_MESSAGE_EOS:
>       g_print ("The source got dry\n");
>       source = gst_bin_get_by_name (GST_BIN (data->sink),
> "testsource");
>       gst_app_src_end_of_stream (GST_APP_SRC (source));
>       break;
>     case GST_MESSAGE_ERROR:
>       g_print ("Received error\n");
>       g_main_loop_quit (data->loop);
>       break;
>     default:
>       break;
>   }
>   return TRUE;
> }
> 
> /* called when we get a GstMessage from the sink pipeline when we get
> EOS, we
>  * exit the mainloop and this testapp. */
> static gboolean
> on_sink_message (GstBus * bus, GstMessage * message, ProgramData *
> data)
> {
>   /* nil */
>   switch (GST_MESSAGE_TYPE (message)) {
>     case GST_MESSAGE_EOS:
>       g_print ("Finished playback\n");
>       g_main_loop_quit (data->loop);
>       break;
>     case GST_MESSAGE_ERROR:
>       g_print ("Received error\n");
>       g_main_loop_quit (data->loop);
>       break;
>     default:
>       break;
>   }
>   return TRUE;
> }
> 
> int
> main (int argc, char *argv[])
> {
>   gchar *filename = NULL;
>   ProgramData *data = NULL;
>   gchar *string = NULL;
>   GstBus *bus = NULL;
>   GstElement *testsink = NULL;
>   GstElement *testsource = NULL;
> 
>   gst_init (&argc, &argv);
> 
>   if (argc == 2)
>     filename = g_strdup (argv[1]);
>   else
>     filename = g_strdup ("/usr/share/sounds/ekiga/ring.wav");
> 
>   data = g_new0 (ProgramData, 1);
> 
>   data->loop = g_main_loop_new (NULL, FALSE);
> 
>   /* setting up source pipeline, we read from a file and convert to
> our desired
>    * caps. */
>   string =
>       g_strdup_printf
>       ("videotestsrc !
> video/x-raw-yuv,width=640,height=480,framerate=15/1 ! appsink caps=\"%
> s\" name=testsink",
>       filename, video_caps);
>   g_free (filename);
>   data->source = gst_parse_launch (string, NULL);
>   g_free (string);
> 
>   if (data->source == NULL) {
>     g_print ("Bad source\n");
>     return -1;
>   }
> 
>   /* to be notified of messages from this pipeline, mostly EOS */
>   bus = gst_element_get_bus (data->source);
>   gst_bus_add_watch (bus, (GstBusFunc) on_source_message, data);
>   gst_object_unref (bus);
> 
>   /* we use appsink in push mode, it sends us a signal when data is
> available
>    * and we pull out the data in the signal callback. We want the
> appsink to
>    * push as fast as it can, hence the sync=false */
>   testsink = gst_bin_get_by_name (GST_BIN (data->source), "testsink");
>   g_object_set (G_OBJECT (testsink), "emit-signals", TRUE, "sync",
> FALSE, NULL);
>   g_signal_connect (testsink, "new-buffer",
>       G_CALLBACK (on_new_buffer_from_source), data);
>   gst_object_unref (testsink);
> 
>   /* setting up sink pipeline, we push audio data into this pipeline
> that will
>    * then play it back using the default audio sink. We have no
> blocking
>    * behaviour on the src which means that we will push the entire
> file into
>    * memory. */
>   string =
>       g_strdup_printf ("appsrc name=testsource caps=\"%s\" !
> xvimagesink",
>       video_caps);
>   data->sink = gst_parse_launch (string, NULL);
>   g_free (string);
> 
>   if (data->sink == NULL) {
>     g_print ("Bad sink\n");
>     return -1;
>   }
> 
>   testsource = gst_bin_get_by_name (GST_BIN (data->sink),
> "testsource");
>   /* configure for time-based format */
>   g_object_set (testsource, "format", GST_FORMAT_TIME, NULL);
>   /* uncomment the next line to block when appsrc has buffered enough
> */
>   /* g_object_set (testsource, "block", TRUE, NULL); */
>   gst_object_unref (testsource);
> 
>   bus = gst_element_get_bus (data->sink);
>   gst_bus_add_watch (bus, (GstBusFunc) on_sink_message, data);
>   gst_object_unref (bus);
> 
>   /* launching things */
>   gst_element_set_state (data->sink, GST_STATE_PLAYING);
>   gst_element_set_state (data->source, GST_STATE_PLAYING);
> 
>   /* let's run !, this loop will quit when the sink pipeline goes EOS
> or when an
>    * error occurs in the source or sink pipelines. */
>   g_print ("Let's run!\n");
>   g_main_loop_run (data->loop);
>   g_print ("Going out\n");
> 
>   gst_element_set_state (data->source, GST_STATE_NULL);
>   gst_element_set_state (data->sink, GST_STATE_NULL);
> 
>   gst_object_unref (data->source);
>   gst_object_unref (data->sink);
>   g_main_loop_unref (data->loop);
>   g_free (data);
> 
>   return 0;
> }
> 
> 
> -- 
>  Mattias Frank Barthel      
>  Vía Augusta 177 08021 Barcelona (Spain)
>  Video & Multimedia technologies
>  TELEFÓNICA R&D
>  T. 0034-93-365-3309
> ------------------------------------------------------------------------------
> Crystal Reports &#45; New Free Runtime and 30 Day Trial
> Check out the new simplified licensign option that enables unlimited
> royalty&#45;free distribution of the report engine for externally facing 
> server and web deployment.
> http://p.sf.net/sfu/businessobjects
> _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel