Hi there,
I am playing around with GStreamer and by coincident I noticed that
when streaming from the Danish national radio's website I got poor
quality audio.
It seems that this is caused by the fact that 'asfdemux' element
chooses the wrong (lowest bitrate) stream.
Now I am wondering if I am using the elements in a wrong way, or if
there indeed is a problem in the asfdemux element.
It can easily be reproduced with gst-launch:
gst-launch mmssrc
location="mmsh://wmscr1.dr.dk/e02ch03m?wmcontentbitrate=300000" !
asfdemux ! ffdec_wmav2 ! pulsesink
When debugging of the asfdemux node is enabled, the following can be
located in the output:
asfdemux gstasfdemux.c:2534:gst_asf_demux_process_bitrate_props_object:
bitrate[1] = 128904
asfdemux gstasfdemux.c:2534:gst_asf_demux_process_bitrate_props_object:
bitrate[2] = 64896
asfdemux gstasfdemux.c:2534:gst_asf_demux_process_bitrate_props_object:
bitrate[3] = 32844
and a bit further down:
asfdemux gstasfdemux.c:1819:gst_asf_demux_activate_stream:<asfdemux0>
Activating stream 3, pad audio_00, caps audio/x-wma... [cut]
There does not seem to be any interesting properties that can be set
on asfdemux or mmssrc. I though "connection-speed" of mmssrc migth be
interesting, but it seems to be set to the maximum value to begin
with. I tried to look into the asfdemux source code, but it is a bit
overwhelming. So
before I waste a lot of time in there I want to be sure that I'm not
rediscovering what others already know :-)
Oh, and by the way, I am using GStreamer 0.10.21, plugins-ugly
0.10.9, plugins-bad 0.10.8. The default versions in the Ubuntu 8.10
repository.
Hope that someone can point me in the right direction to solve this problem.
Cheers,
Paul
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