<div dir="ltr">Hello again.<div><br></div><div>Tim's solutions have worked great when we were using FFSERVER to re-stream the video.</div><div>However we now are implementing GST_RTPS_SERVER and the Android Player is the only one not playing the video. FFPlay, VLC, and a Windows Gstreamer player can all reproduce the stream that is represented by the uri rtsp://<a href="http://127.0.0.1:8554/test">127.0.0.1:8554/test</a> (the FFSERVER uri had a .sdp )</div><div><br></div><div>I can see the Android Player requisition on the RTSP Server terminal:</div><div><br></div><div>(lt-test-launch:28849): GLib-GObject-WARNING **: g_object_get_valist: object class 'GstIdentity' has no property named 'pt'<br></div><div><br></div><div>But still no playing...</div><div><br></div><div>Should I add another parameter to the android player? Or should I change the way the RTSP Server builds the URI?</div><div><br></div><div>Thank you.</div></div><div class="gmail_extra"><br><div class="gmail_quote">On Tue, Sep 26, 2017 at 10:28 AM, Olivier Crête <span dir="ltr"><<a href="mailto:olivier.crete@collabora.com" target="_blank">olivier.crete@collabora.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div>Hi,</div><div><br></div><div>The two seconds are from rtspsrc, you can set the "latency" property to something reasonable like 200ms by connecting to the "source-setup" signal on playbin and setting the latency property to 200 if it's rtspsrc.</div><div><br></div><div>You can verify the automatically calculated latency by setting GST_DEBUG=bin:4</div><div><br></div><div>Setting the pipeline latency manually will indeed produce problems if you set a lower value than the minimal one.</div><div><br></div><div>Olivier</div><span class=""><div><br></div><div>On Wed, 2017-09-20 at 14:13 -0300, Azure1 Sureale wrote:</div></span><blockquote type="cite" style="margin:0 0 0 .8ex;border-left:2px #729fcf solid;padding-left:1ex"><div><div class="h5"><div dir="ltr">Hello, I'm trying to have an Android Player receiving a RTSP stream from the internet and play it with very low latency. <div>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).</div><div><br></div><div>With futher investigation we realized we could change the latency to 0 in the pipeline using this code:</div><div><br></div><div>******************************<wbr>******************************<wbr>****</div><div><div>data->pipeline = gst_parse_launch("playbin", &error);</div><div> gst_pipeline_set_latency(data-<wbr>>pipeline,0); //This is what I wrote</div><div><br></div><div> if (error) {</div><div> gchar *message = g_strdup_printf("Unable to build pipeline: %s", error->message);</div><div> g_clear_error (&error);</div><div> set_ui_message(message, data);</div><div> g_free (message);</div><div> return NULL;</div><div> }</div><br class="m_-2069707513643876019gmail-Apple-interchange-newline">******************************<wbr>******************************<wbr>****</div><div><br></div><div>Now the stream is strutting, alas dropping frames. However the stream is faster than the base code.</div><div>Based on other questions, I tried to add rtpjitterbuffer into the pipeline but seems that is not that simple.</div><div><br></div><div>How can I construct or change the pipeline to have low latency and no strutting?</div></div>
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