Hi,
I have pipeline which sends audio from .wav file via RTP to VoIP server (with help of rtpbin element), and at the same time receives audio from it and saves it to another wav file. I wanted to measure delay introduced by VoIP server, by saving both input audio and output audio to separate channels of .wav file. To do this, I split decoded stream from .wav file into two using tee, and then send it to interleave element (plus necessary audioconvert/audiopresample/capsfilter) as channel 1. Channel 2 gets decoded rtp stream. When I opened recorded file, I found that delay is about 5 secs, plus similar part of initial audio on channel 1 was truncated. GStreamer itself introduces small delay - on all 3 paths (filesrc->udpsink, udpsrc->filesink, filesrc->filesink) it is below 1ms. I also decoded captured RTP streams in Wireshark, and delay between them was 23ms. So things clearly do not sum up. Looks that this is synchronization problem. I added sync=TRUE to filesink, what cut down about 1 sec. I also added audiomixer + audiotestsrc as silence generator after rtpbin and decoders to always have this part of pipeline connected - before I had to wait for signal from rtpbin that it added corresponding output pad. Now delay in recorded file is about 2 secs, what still is way too big. Looks that my streams are still not properly synchronized. What am I missing here? I also still see that initial audio truncation. I stop pipeline by changing state to NULL, so file is not closes properly and some trailing audio is lost. Is is possible that initial part can be lost too because of this? BTW, I have just noticed that GStreamer provides interleave and audiointerleave elements, and it seems that both of them do the same thing. Why 2 similar elements exists? Regards, Daniel _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
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interleave element is deprecated and you should use audiointerleave instead.
audiointerleave element is written on top of GstAggregator which works very well for live inputs unlike the interleave element ~BO -- Sent from: http://gstreamer-devel.966125.n4.nabble.com/ _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
W dniu 2018-05-17 13:19, Baby Octopus napisał(a):
> interleave element is deprecated and you should use audiointerleave > instead. > audiointerleave element is written on top of GstAggregator which works > very > well for live inputs unlike the interleave element > > ~BO > > > > -- > Sent from: http://gstreamer-devel.966125.n4.nabble.com/ > _______________________________________________ > gstreamer-devel mailing list > [hidden email] > https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel Thanks! audiointerleave element works much better, looks that both streams are synchronized now. It also seem that delay introduces by jitterbuffer is removed. However beginning of stream coming from tee element is still truncated - about 3.2 secs of initial audio is replaced with silence. I have queues added before both sinks of audiointerleave, so it is surprising to me. What am I doing wrong? Regards, Daniel _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
Le jeudi 17 mai 2018 à 18:22 +0200, [hidden email] a
écrit : > Thanks! audiointerleave element works much better, looks that both > streams are synchronized now. It also seem that delay introduces by > jitterbuffer is removed. > > However beginning of stream coming from tee element is still truncated - > about 3.2 secs of initial audio is replaced with silence. I have queues > added before both sinks of audiointerleave, so it is surprising to me. > What am I doing wrong? I'm not sure, maybe because of the start time. You may want to try with start-time-selection=1. Sometimes adding some latency (See "latency" property" resolved some timing issues). Nicolas _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel signature.asc (201 bytes) Download Attachment |
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