<div dir="ltr"><div><div><div>Thank you very much for your reply. </div>It is working fine now, it was silly mistake from my side. </div>Regards </div>Jyoti </div><div class="gmail_extra"> <div class="gmail_quote"> On Thu, Aug 29, 2013 at 1:15 PM, Wim Taymans <<a href="mailto:wim.taymans@gmail.com" target="_blank">wim.taymans@gmail.com</a>> wrote: <blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"> <div class="HOEnZb"><div class="h5">On 08/29/2013 08:25 AM, jyoti kulkarni wrote: <blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"> Hello All, My pipeline is: gst-launch-1.0 -v appsrc ! qtdemux name=d d.audio_0 ! queue ! avdec_aac ! appsink d.video_0 ! queue ! avdec_h264 ! appsink and when ever i get a new decoded audio/video buffer i put that in appsink. Then i use that same buffer to file write. I do it in following way: For Audio: (here data.audio_sink is nothing but appsink) g_object_set (data.audio_sink, "emit-signals", TRUE, NULL); g_signal_connect (data.audio_sink, "new-sample", G_CALLBACK (new_audio_buffer), &data); static void new_audio_buffer (GstElement *sink,CustomData *data) { GstBuffer *buffer; GstMapInfo map; GstSample *sample; //gsize size; g_signal_emit_by_name (data->audio_sink, "pull-sample", &sample,NULL); if (sample) { buffer = gst_sample_get_buffer (sample); gst_buffer_map (buffer, &map, GST_MAP_READ); g_print("\n here size=%d\n",map.size); fwrite(map.data,1,map.size,data->audio_file); gst_buffer_unmap (buffer,&map); gst_sample_unref(sample); } } AND similar kind of code for video appsink. But the problem is only single buffer data is written to the file there after gstreamer returns with a "General Stream Error" The same application worked with gstreamer-0.10, where i used new_buffer signal instead of new_sample and corresponding code for user defined function was: g_object_set (data.audio_sink, "emit-signals", TRUE, NULL); g_signal_connect (data.audio_sink, "new-buffer", G_CALLBACK (new_audio_buffer), &data); static void new_audio_buffer (GstElement *sink, CustomData *data) { </blockquote> </div></div> You callback is wrong, it should return a GstFlowReturn, see gst-inspect-1.0 appsink for the signature of the callback. Wim <blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div class="im"> GstBuffer *buffer; g_signal_emit_by_name (sink, "pull-buffer", &buffer); if (buffer) { gint size=GST_BUFFER_SIZE(buffer); g_print("\nsize=%d \n",size); fwrite(GST_BUFFER_DATA(buffer), 1, GST_BUFFER_SIZE(buffer),data->audio_file); gst_buffer_unref (buffer); } } Here all the buffers data written successfully to the file until the source file reaches end of file. Please help with above concern. Thanks & Regards Jyoti </div> _______________________________________________ gstreamer-devel mailing list <a href="mailto:gstreamer-devel@lists.freedesktop.org" target="_blank">gstreamer-devel@lists.freedesktop.org</a> <a href="http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel" target="_blank">http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel</a> </blockquote> _______________________________________________ gstreamer-devel mailing list <a href="mailto:gstreamer-devel@lists.freedesktop.org" target="_blank">gstreamer-devel@lists.freedesktop.org</a> <a href="http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel" target="_blank">http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel</a> </blockquote></div> </div>