<html><head><meta http-equiv="Content-Type" content="text/html charset=utf-8"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class="">Hi Zheng<div class=""> </div><div class="">Whenever possible, I use <a href="https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstParse.html#gst-parse-launch" class="">gst_parse_launch</a> to avoid these types of complications. This is what gst-launch uses underneath, so if it works on one it should work on the other.</div><div class=""> </div><div class="">Specifically in your example:</div><div class=""> </div><div class="">int startPipeline(const char *vs_location) { GstElement *pipeline; GstBus *bus; guint bus_watch_id; gchar *pipe_desc;</div><div class=""> GError *error = NULL;</div><div class=""> </div><div class=""> gst_init(NULL, NULL);</div><div class=""> </div><div class=""> pipe_desc = g_strdup_printf("rtspsrc port-range=5000-5100 location=%s latency=0 ! decodebin ! videoconvert ! video/x-raw,width=640,height=480,format=YUY2 ! autovideosink”, vs_location);</div><div class=""> pipeline = gst_parse_launch (pipe_desc, &error);</div><div class=""> //Handle errors here</div><div class=""> g_free (pipe_desc);</div><div class=""> bus = gst_pipeline_get_bus(GST_PIPELINE(pipeline)); bus_watch_id = gst_bus_add_watch(bus, bus_callback, NULL); gst_object_unref(bus); gst_element_set_state(pipeline, GST_STATE_PLAYING); loop = g_main_loop_new(NULL, FALSE); g_main_loop_run(loop); // Clean up. gst_element_set_state(pipeline, GST_STATE_NULL); gst_object_unref(pipeline); g_source_remove(bus_watch_id); g_main_loop_unref(loop); return 0; }</div><div class=""> <div class=""> <div class=""><div class="">—</div><div class="">Michael Gruner <<a href="mailto:michael.gruner@ridgerun.com" class="">michael.gruner@ridgerun.com</a>></div><div class="">Embedded Linux and GStreamer solutions</div><div class="">RidgeRun Engineering</div><div class="">Contact Us - <a href="http://www.ridgerun.com/#!contact/c3vn" class="">http://www.ridgerun.com/#!contact/c3vn</a></div></div> </div> <div><blockquote type="cite" class=""><div class="">On Jun 16, 2016, at 19:45, Zheng <<a href="mailto:superxingzheng@gmail.com" class="">superxingzheng@gmail.com</a>> wrote:</div> <div class=""><div class="">*Thang you in advance!* *This is the working gst-launch-1.0 pipeline.* gst-launch-1.0 rtspsrc port-range=5000-5100 location=<a href="rtsp://192.168.1.10:8555/right" class="">rtsp://192.168.1.10:8555/right</a> latency=0 ! decodebin ! videoconvert ! video/x-raw,width=640,height=480,format=YUY2 ! autovideosink *The following is the NOT working C function.* int startPipeline(const char *vs_location) { GstElement *pipeline; GstBus *bus; guint bus_watch_id; gst_init(NULL, NULL); pipeline = gst_pipeline_new("my_pipeline"); bus = gst_pipeline_get_bus(GST_PIPELINE(pipeline)); bus_watch_id = gst_bus_add_watch(bus, bus_callback, NULL); gst_object_unref(bus); GstElement *source, *dec_bin, *v_conv, *sink; source = gst_element_factory_make("rtspsrc", "source"); dec_bin = gst_element_factory_make("decodebin", "dec_bin"); v_conv = gst_element_factory_make("videoconvert", "v_con"); sink = gst_element_factory_make("autovideosink", "sink"); gst_bin_add_many(GST_BIN(pipeline), source, dec_bin, v_conv, sink, NULL); // Set element properties. g_object_set(G_OBJECT(source), "port-range", "5000-5100", NULL); g_object_set(G_OBJECT(source), "location", vs_location, NULL); g_object_set(G_OBJECT(source), "latency", 0, NULL); gst_element_link_many(source, dec_bin, v_conv, NULL); GstCaps *caps; caps = gst_caps_new_simple( "video/x-raw", "width", G_TYPE_INT, 640, "height", G_TYPE_INT, 480, "format", G_TYPE_STRING, "YUY2", NULL); gst_element_link_filtered(v_conv, sink, caps); gst_caps_unref(caps); gst_element_set_state(pipeline, GST_STATE_PLAYING); loop = g_main_loop_new(NULL, FALSE); g_main_loop_run(loop); // Clean up. gst_element_set_state(pipeline, GST_STATE_NULL); gst_object_unref(pipeline); g_source_remove(bus_watch_id); g_main_loop_unref(loop); return 0; } *Then the Errors are here.* 0:00:00.442295071 16407 0x7f007c01e940 FIXME default gstutils.c:3825:gst_pad_create_stream_id_internal:<fakesrc0:src> Creating random stream-id, consider implementing a deterministic way of creating a stream-id 0:00:00.455584401 16407 0x7f006c0025e0 FIXME rtpjitterbuffer gstrtpjitterbuffer.c:1395:gst_jitter_buffer_sink_parse_caps:<rtpjitterbuffer0> Unsupported timestamp reference clock 0:00:00.455616947 16407 0x7f006c0025e0 FIXME rtpjitterbuffer gstrtpjitterbuffer.c:1403:gst_jitter_buffer_sink_parse_caps:<rtpjitterbuffer0> Unsupported media clock 0:00:00.457236636 16407 0x7f006c0025e0 WARN basesrc gstbasesrc.c:2947:gst_base_src_loop:<udpsrc0> error: Internal data flow error. 0:00:00.457268553 16407 0x7f006c0025e0 WARN basesrc gstbasesrc.c:2947:gst_base_src_loop:<udpsrc0> error: streaming task paused, reason not-linked (-1) Error: Internal data flow error. Debug: gstbasesrc.c(2947): gst_base_src_loop (): /GstPipeline:my_pipeline/GstRTSPSrc:source/GstUDPSrc:udpsrc0: streaming task paused, reason not-linked (-1) 0:00:00.458046376 16407 0x10af5e0 WARN rtspsrc gstrtspsrc.c:5520:gst_rtspsrc_try_send:<source> receive interrupted 0:00:00.458083322 16407 0x10af5e0 WARN rtspsrc gstrtspsrc.c:7546:gst_rtspsrc_pause:<source> PAUSE interrupted -- View this message in context: <a href="http://gstreamer-devel.966125.n4.nabble.com/Simple-RTSP-Pipeline-Works-with-gst-launch-But-Not-with-API-tp4678108.html" class="">http://gstreamer-devel.966125.n4.nabble.com/Simple-RTSP-Pipeline-Works-with-gst-launch-But-Not-with-API-tp4678108.html</a> Sent from the GStreamer-devel mailing list archive at <a href="http://nabble.com" class="">Nabble.com</a>. _______________________________________________ gstreamer-devel mailing list <a href="mailto:gstreamer-devel@lists.freedesktop.org" class="">gstreamer-devel@lists.freedesktop.org</a> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel </div></div></blockquote></div> </div></body></html>