<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> </head> <body bgcolor="#ffffff" text="#000000"> 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). 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; } <pre class="moz-signature" cols="800">-- Mattias Frank Barthel Vía Augusta 177 08021 Barcelona (Spain) Video & Multimedia technologies TELEFÓNICA R&D T. 0034-93-365-3309 </pre> </body> </html>