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Bug 752716 - audio rtp L16 streaming to alsa
audio rtp L16 streaming to alsa
Status: RESOLVED OBSOLETE
Product: GStreamer
Classification: Platform
Component: gst-plugins-good
0.10.x
Other Linux
: Normal normal
: NONE
Assigned To: GStreamer Maintainers
GStreamer Maintainers
Depends on:
Blocks:
 
 
Reported: 2015-07-22 12:45 UTC by Roland H.
Modified: 2015-07-22 13:07 UTC
See Also:
GNOME target: ---
GNOME version: ---



Description Roland H. 2015-07-22 12:45:10 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
Comment 1 Tim-Philipp Müller 2015-07-22 13:07:29 UTC
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).