<div dir="ltr">For mp3 it works fine. For wav and raw pcm files  program never get signal handoff. To play wav file I used dynamic pad and for pcm doesn't need pad. Is it possible to play binary data using this approach? If so what would be the decoder elements. I have looked into appsrc plugin . I ran the example given as part of the plugin. Audio is not coming. Thanks, Akbar <div class="gmail_quote">On Sat, Jul 12, 2008 at 3:33 PM, Stefan Kost <<a href="mailto:ensonic@hora-obscura.de">ensonic@hora-obscura.de</a>> wrote: <blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">hi, Akbar Basha schrieb:<div class="Ih2E3d"> <blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"> Hi Stefan, Thanks for the response . </blockquote> </div> Would you please post to the list.<div><div></div><div class="Wj3C7c"> <blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"> I tried the same. But could not produce the result. Please find the code. static void cb_handoff (GstElement *fakesrc,        GstBuffer  *buffer,         gpointer    user_data) {    /* Clip start and end */  data = (guint8 *) g_malloc (3000);  GST_BUFFER_SIZE (buffer) = 3000;  GST_BUFFER_DATA (buffer) = GST_BUFFER_MALLOCDATA (buffer) = data;    FILE* fp = fopen("vertigo.mp3","rb");  if(fp == NULL)   {    printf( " File is not opened \n");    return;   }  fread(data,3000,1,fp);    fclose(fp);  } gint main (gint   argc,      gchar *argv[]) {  GstElement *pipeline, *fakesrc, *flt, *conv, *audiosink;  GMainLoop *loop;  /* init GStreamer */  gst_init (&argc, &argv);  loop = g_main_loop_new (NULL, FALSE);  /* setup pipeline */  pipeline = gst_pipeline_new ("pipeline");  fakesrc = gst_element_factory_make ("fakesrc", "source");  flt = gst_element_factory_make ("capsfilter", "flt");  conv = gst_element_factory_make ("mad", "conv");  audiosink = gst_element_factory_make ("alsasink", "audiosink");  /* setup */  g_object_set (G_OBJECT (flt), "caps",          gst_caps_new_simple("audio/x-raw-int",                "channels", G_TYPE_INT, 2,                "rate", G_TYPE_INT, 32000,               "depth", G_TYPE_INT, 16, NULL), NULL); </blockquote> </div></div> This is obviously wrong. You load an mp3 and not raw audio data. The capsfilter needs to tell that. But in your case you would not even need one.<div class="Ih2E3d"> <blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">  gst_bin_add_many (GST_BIN (pipeline), fakesrc, flt,conv, audiosink, NULL);  gst_element_link_many (fakesrc, flt,conv, audiosink, NULL);  /* setup fake source */  g_object_set (G_OBJECT (fakesrc),"signal-handoffs", TRUE,NULL);  g_signal_connect (fakesrc, "handoff", G_CALLBACK (cb_handoff), NULL);  /* play */  gst_element_set_state (pipeline, GST_STATE_PLAYING);  g_main_loop_run (loop);  /* clean up */  gst_element_set_state (pipeline, GST_STATE_NULL);  gst_object_unref (GST_OBJECT (pipeline));  return 0; } Even if I set using memset . Audio is not coming. </blockquote> </div> What happens? Stefan <blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div class="Ih2E3d"> how to proceed in the case pad is required  i.e for wav files. Regards, Akbar </div><div><div></div><div class="Wj3C7c"> On Wed, Jul 9, 2008 at 11:53 PM, Stefan Kost <<a href="mailto:ensonic@hora-obscura.de" target="_blank">ensonic@hora-obscura.de</a> <mailto:<a href="mailto:ensonic@hora-obscura.de" target="_blank">ensonic@hora-obscura.de</a>>> wrote:    Akbar Basha schrieb:        Hi,        I would like to play the buffer , which is filled with any audio        file data.        Does gstreamer provides any mechanism to play.    if you have the whole bufer in memory, use a fakesrc with    signal-handoffs=TRUE and connect to handoff signal. In the handoff    signal you put the pointer to your data into the GST_BUFFER_DATA,    set the correct GST_BUFFER_SIZE and clear GST_BUFFER_MALLOC_DATA (if    it was previously set g_free() the previous content).    You should used a capsfilter after fakesrc and set the format of    your sample on the capsfilter caps.    Its sort of a hack, but works fine.    Stefan        Thanks in advance.        Regards,        Akbar        ------------------------------------------------------------------------        -------------------------------------------------------------------------        Sponsored by: SourceForge.net Community Choice Awards: VOTE NOW!        Studies have shown that voting for your favorite open source        project,        along with a healthy diet, reduces your potential for chronic        lameness        and boredom. 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