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<p class="MsoNormal"><span lang="EN-US" style="font-size:10.0pt;font-family:"Arial","sans-serif"">Hi<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US" style="font-size:10.0pt;font-family:"Arial","sans-serif"">I have a similar pipeline based on the rtsp appsrc example. I use the client_connected and client_closed callbacks to start and stop my pipeline. I can attach
and detach clients more than once.<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">Von:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif""> gstreamer-devel [mailto:gstreamer-devel-bounces@lists.freedesktop.org]
<b>Im Auftrag von </b>Serj TorresSoldado<br>
<b>Gesendet:</b> Dienstag, 29. März 2016 11:34<br>
<b>An:</b> Discussion of the development of and with GStreamer<br>
<b>Betreff:</b> Re: Video corruption in rtsp server with appsink->appsrc<o:p></o:p></span></p>
<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">I have just noticed this happens with the gst-rtsp-server examples as well.<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">I am using the 1.6 branch.<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">On 28 March 2016 at 21:50, Serj TorresSoldado <<a href="mailto:torres.soldado@gmail.com" target="_blank">torres.soldado@gmail.com</a>> wrote:<o:p></o:p></p>
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<p class="MsoNormal">Hi All,<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal"> I am doing a sink -> src push. The sink is in a separate pipeline and the src is created by gst-rtsp-server when creating the media pipeline.<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal"> I have tried both using a pull (need-data signal) and push (new-sample) methods and the result is the same.<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal"> I am copying the buffer from the sink to the source. I am setting the PTS on the copied buffer otherwise after the first client disconnects I am unable to connect again.<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal"> GST_BUFFER_PTS(bufcpy) = client->timestamp_;<o:p></o:p></p>
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<p class="MsoNormal"> client->timestamp_ += GST_BUFFER_DURATION(bufcpy);<o:p></o:p></p>
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<p class="MsoNormal"> gst_app_src_push_buffer(client->appsrc_, bufcpy);<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal"> When cofiguring the client I have "played" with the following appsrc properties but they don't seem to make a difference:<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal"> gst_util_set_object_arg(reinterpret_cast<GObject*>(appsrc.Get()), "format", "time");<o:p></o:p></p>
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<p class="MsoNormal"> g_object_set(reinterpret_cast<GObject*>(appsrc.Get()), "block", false, nullptr);<o:p></o:p></p>
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<p class="MsoNormal"> g_object_set(reinterpret_cast<GObject*>(appsrc.Get()), "is-live", true, nullptr);<o:p></o:p></p>
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<p class="MsoNormal"> g_object_set(reinterpret_cast<GObject*>(appsrc.Get()), "do-timestamp", true, nullptr);<o:p></o:p></p>
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<p class="MsoNormal">// g_object_set(reinterpret_cast<GObject*>(appsrc.Get()), "stream-type", "random-access", nullptr);<o:p></o:p></p>
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<p class="MsoNormal">// g_object_set(reinterpret_cast<GObject*>(appsrc.Get()), "max-bytes", 1000, nullptr);<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">Any help would be awesome, thanks.<o:p></o:p></p>
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<p class="MsoNormal"><span style="color:#888888"><o:p> </o:p></span></p>
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<p class="MsoNormal"><span style="color:#888888">Serj<o:p></o:p></span></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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