Hi, <div class="gmail_quote">On Wed, Oct 24, 2012 at 6:26 PM, rasnaut <<a href="mailto:rasnaut@gmail.com" target="_blank">rasnaut@gmail.com</a>> wrote: <blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"> Hi to All! I working with gstreamer-1.0 and try use fakesrc for work with audiostream. For start trying I copy only 0 in buffer (I used MapInfo structure). So, my code: </blockquote><div> Have you tried to use audiotestsrc instead of fakesrc ? (you will not need the audiorate element) <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-plugins/html/gst-plugins-base-plugins-audiotestsrc.html">http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-plugins/html/gst-plugins-base-plugins-audiotestsrc.html</a> Best regards, Tiago Katcipis </div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"> main() { gst_init (NULL,NULL); loop = g_main_loop_new (NULL, FALSE); /* setup pipeline */ pipeline = gst_pipeline_new ("pipeline"); g_assert(pipeline); fakesrc = gst_element_factory_make ("fakesrc", "source"); g_assert(fakesrc); flt = gst_element_factory_make ("capsfilter", "flt"); g_assert(flt); rate = gst_element_factory_make ("audiorate", "rate"); g_assert(rate); conv = gst_element_factory_make ("audioconvert", "conv"); g_assert(conv); audiosink = gst_element_factory_make ("alsasink", "asink"); g_assert(videosink); /* setup */ g_object_set (G_OBJECT (flt), "caps", gst_caps_new_simple ("audio/x-raw", "format",G_TYPE_STRING,"S16LE", "rate", G_TYPE_INT,16000, "channels", G_TYPE_INT, 1, NULL), NULL); gst_bin_add_many (GST_BIN (pipeline), fakesrc, flt,rate, conv, audiosink, NULL); if(!gst_element_link_many (fakesrc, flt, rate, conv, audiosink, NULL)) { g_error("Link Error\n"); } /* setup fake source */ g_object_set (G_OBJECT (fakesrc), "signal-handoffs", TRUE, "sizemax", 16000, "sizetype", 2, 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)); g_main_loop_unref (loop); } So, when I start my programm apear next error: GStreamer-CRITICAL **: gst_segment_to_running_time: assertion `segment->format == format' failed When I changed alsasink on fakesink, all work. Maybe it's because my audiocard is working with another details: format = SL32_LE rate = 48000 channels = 2 I tryed change my details: g_object_set (G_OBJECT (flt), "caps", gst_caps_new_simple ("audio/x-raw", "format",G_TYPE_STRING,"S32LE", "rate", G_TYPE_INT,48000, "channels", G_TYPE_INT, 2, NULL), NULL); g_object_set (G_OBJECT (fakesrc), "signal-handoffs", TRUE, "sizemax", 48000*2, "sizetype", 8, NULL); but error still stayed. Anybody know, what I do wrong? -- View this message in context: <a href="http://gstreamer-devel.966125.n4.nabble.com/fakesrc-for-audio-stream-tp4656701.html" target="_blank">http://gstreamer-devel.966125.n4.nabble.com/fakesrc-for-audio-stream-tp4656701.html</a> Sent from the GStreamer-devel mailing list archive at Nabble.com. _______________________________________________ gstreamer-devel mailing list <a href="mailto:gstreamer-devel@lists.freedesktop.org">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>