<html> <head> <meta http-equiv="content-type" content="text/html; charset=UTF-8"> </head> <body> Hello I'm a software programer from Barcelona and my company develops professional video servers for broadcasting. We use propietary SDKs for managing video and audio, like MatroxDSX, Medialooks, BlackMagicSDK, ... For a recent project we are studying the possibility to use GStreamer form sending an SRT stream, but the resulting stream in some receivers, no all, like VLC, drops audio samples and I don't know the reason. If I use this pipe, the resulting stream is Ok. Players like VLC are able to connect them using <a href="srt://127.0.0.1:5011?mode=caller" class="moz-txt-link-freetext">srt://127.0.0.1:5011?mode=caller</a> and the works fine, with no errors and valid video and audio data gst-launch-1.0 uridecodebin uri="<a>file:///C:/temp/AT_0007</a> Corzo emboscado.mxf" name=decode ! videoconvert ! x264enc ! queue ! mpegtsmux name=mux ! queue ! srtsink uri=<a href="srt://127.0.0.1:5011?mode=listener" class="moz-txt-link-freetext">srt://127.0.0.1:5011?mode=listener</a> decode. ! audioconvert ! avenc_aac ! queue ! mux. But if I use appsrc the problem appears in some players, like VLC, but there are no errors in the log window appsrc is-live=true do-timestamp=true ! videoconvert ! x264enc ! queue ! mpegtsmux name=mux ! queue ! srtsink uri=<a href="srt://127.0.0.1:5011?mode=listener" class="moz-txt-link-freetext">srt://127.0.0.1:5011?mode=listener</a> appsrc is-live=true do-timestamp=true ! audioconvert ! avenc_aac ! queue ! mux. I have done multiple tests with no success, removing queues, using UDP instead of SRT, saving to disk the trasport stream and playing back using VLC... The "need-data" callbacks for audio and video are very easy The always push a buffer of one frame for video and the corresponding audio data for audio. The timestamps are ok. If instead of send the data to a muxer we change the pipe to render the incoming video and audio, there are no problems, and the audio sounds ok. It really looks like a timestamp problem, but I can't find the solution This is the "need-data code" bool want_video = false; void cb_need_video_data(GstElement *src, guint unused_size, gpointer user_data) { want_video = true; return; } void write_video_data(const void *data, int data_size) { if(!want_video) return false; want_video = false; // Fetch num den factors GstStructure *s = gst_caps_get_structure(m_vCaps, 0); gint num = 0, den = 0; gst_structure_get_fraction(s, "framerate", &num, &den); GstBuffer *buffer = gst_buffer_new_wrapped_full(GST_MEMORY_FLAG_READONLY, (gpointer) data, data_size, 0, data_size, NULL, NULL); static GstClockTime timestamp = 0; GST_BUFFER_PTS(buffer) = timestamp; GST_BUFFER_DURATION(buffer) = gst_util_uint64_scale_int(GST_SECOND, den, num); // GST_BUFFER_DTS(buffer) = timestamp - GST_BUFFER_DURATION(buffer); timestamp += GST_BUFFER_DURATION(buffer); GstFlowReturn ret = gst_app_src_push_buffer((GstAppSrc *) vAppSrc, buffer); } And the same for audio. The incoming data comes from our video server Can anybody help me? I'm working on Windows with the last GStreamer version. </body> </html>