GNOME Bugzilla – Bug 752716
audio rtp L16 streaming to alsa
Last modified: 2015-07-22 13:07:29 UTC
I want to stream audio to a alsa device, which is capable of playing 16 bit, one channel, signed audio at a rate of 6000. This is working for a test pipeline: gst-launch -vvv filesrc location=tst.wav ! wavparse ! alsasink device=plughw:1,0 pipeline return: alsasink0.GstPad:sink: caps = audio/x-raw-int, endianness=(int)1234, channels=(int)1, width=(int)16, depth=(int)16, signed=(boolean)true, rate=(int)8000 With the following pipeling sound is played scrambled: gst-launch -vvv audiotestsrc ! audioresample ! "audio/x-raw-int,width=16,channels=1,rate=8000" ! alsasink device=hw:1,0 sync=false At the end I need a pipeline like this, but this also does not work (audio output is understandable very bad, has a lot of diturbances): gst-launch -vvv udpsrc port=5006 caps="application/x-rtp,clock-rate=44100,payload=11" ! rtpL16depay ! audioconvert ! "audio/x-raw-int,endianness=1234" ! audioresample quality=10 ! "audio/x-raw-int,rate=8000,clock-rate=8000" ! alsasink device=plughw:1,0 sync=false An example stream for this pipeline is created using rtp tools: rtpplay.exe -f hu_to_rfsb2.rtpdump -p rtp.type 192.168.2.255/5006
Thanks for the bug report, but GStreamer 0.10 is no longer supported. It's very old and has been unmaintained for years, please us a recent GStreamer 1.x version instead. Please re-open this bug if you still have problems with a recent 1.x version. A couple of comments: you should be using an rtpjitterbuffer before the depayloader, and alsasink sync=true (which is the default).