<div dir="ltr">Thank you Nicolas. Tim has given a similar solution last week.<div> </div><div>We are now facing a new problem. We moved from ffserver to gst-rtps-server. I explained the issue in a previous email.</div><div class="gmail_extra"> <div class="gmail_quote">On Thu, Sep 28, 2017 at 2:39 PM, Nicolas Dufresne <<a href="mailto:nicolas@ndufresne.ca" target="_blank">nicolas@ndufresne.ca</a>> wrote: <blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Le mercredi 20 septembre 2017 à 14:13 -0300, Azure1 Sureale a écrit : > Hello, I'm trying to have an Android Player receiving a RTSP stream > from the internet and play it with very low latency. > I'm using Tutorial 5 from the GStreamer home. Without any change my > stream has a latency about 2 seconds (worst than Vitamio that I was > using previously). This is the default configuration on rtpsrc. Connect to "source-setup" signal on playbin, and then setup a lower latency. Here's a code snipet: static void source_setup_cb (GstElement *playbin, GstElement *source, gpointer user_data) { if (g_str_equal ("GstRTSPSrc", g_type_name (source))) g_object_set (source, "latency", 50, NULL); /* in ms */ } { ... g_signal_connect (pipeline, "source-setup", G_CALLBACK (source_setup_cb), NULL, NULL); ... } > With futher investigation we realized we could change the latency to > 0 in the pipeline using this code: > > **************************************************************** > data->pipeline = gst_parse_launch("playbin", &error); > gst_pipeline_set_latency(data->pipeline,0); //This is what I > wrote That's won't work, since you'll have to display the frame before you even have started decoding them, it make no sense. It's better to configure the element, and let the normal latency workflow run. > > if (error) { > gchar *message = g_strdup_printf("Unable to build pipeline: %s", > error->message); > g_clear_error (&error); > set_ui_message(message, data); > g_free (message); > return NULL; > } > > **************************************************************** > > Now the stream is strutting, alas dropping frames. However the stream > is faster than the base code. > Based on other questions, I tried to add rtpjitterbuffer into the > pipeline but seems that is not that simple. > > How can I construct or change the pipeline to have low latency and no > strutting? The configuration depends on your network, GStreamer don't implement dynamic latency, though you may implement it yourself by warning QoS event about dropped frame (to detect too low latency) and do some adjustment. <div class="m_360833109915751334HOEnZb"><div class="m_360833109915751334h5"> > _______________________________________________ > gstreamer-devel mailing list > <a href="mailto:gstreamer-devel@lists.freedesktop.org" target="_blank">gstreamer-devel@lists.freedesktop.org</a> > <a href="https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel" rel="noreferrer" target="_blank">https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel</a></div></div> _______________________________________________ gstreamer-devel mailing list <a href="mailto:gstreamer-devel@lists.freedesktop.org" target="_blank">gstreamer-devel@lists.freedesktop.org</a> <a href="https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel" rel="noreferrer" target="_blank">https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel</a> </blockquote></div> </div></div>