[gstreamer-bugs] [Bug 439570] New: gst not working with my bluetooth headset
GStreamer (bugzilla.gnome.org)
bugzilla-daemon at bugzilla.gnome.org
Fri May 18 18:52:39 PDT 2007
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GStreamer | don't know | Ver: 0.10.12
Summary: gst not working with my bluetooth headset
Product: GStreamer
Version: 0.10.12
Platform: Other
OS/Version: Linux
Status: UNCONFIRMED
Severity: normal
Priority: Normal
Component: don't know
AssignedTo: gstreamer-bugs at lists.sourceforge.net
ReportedBy: l.bigonville at edpnet.be
QAContact: gstreamer-bugs at lists.sourceforge.net
GNOME version: 2.19/2.20
GNOME milestone: Unspecified
Hi,
I'm currently trying bluetooth-alsa with my bt headset. Some thing like this
arecord -D plug:headset |aplay works perfectly.
So the next step is to make it works with gst.
I try to build a pipeline like this:
gst-launch-0.10 -v alsasrc device=plug:headset ! audioconvert ! audioresample !
alsasink
and I got an error:
Setting pipeline to PAUSED ...
/pipeline0/alsasrc0.src: caps = audio/x-raw-int, endianness=(int)1234,
signed=(boolean)true, width=(int)32, depth=(int)32, rate=(int)44100,
channels=(int)2
Pipeline is live and does not need PREROLL ...
Setting pipeline to PLAYING ...
New clock: GstAudioSrcClock
/pipeline0/audioconvert0.src: caps = audio/x-raw-int, endianness=(int)1234,
signed=(boolean)true, width=(int)32, depth=(int)32, rate=(int)44100,
channels=(int)2
/pipeline0/audioconvert0.sink: caps = audio/x-raw-int, endianness=(int)1234,
signed=(boolean)true, width=(int)32, depth=(int)32, rate=(int)44100,
channels=(int)2
/pipeline0/audioresample0.src: caps = audio/x-raw-int, endianness=(int)1234,
signed=(boolean)true, width=(int)32, depth=(int)32, rate=(int)44100,
channels=(int)2
/pipeline0/audioresample0.sink: caps = audio/x-raw-int, endianness=(int)1234,
signed=(boolean)true, width=(int)32, depth=(int)32, rate=(int)44100,
channels=(int)2
/pipeline0/alsasink0.sink: caps = audio/x-raw-int, endianness=(int)1234,
signed=(boolean)true, width=(int)32, depth=(int)32, rate=(int)44100,
channels=(int)2
WARNING: from element /pipeline0/alsasrc0: Internal GStreamer error: clock
problem. Please file a bug at
http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer.
Additional debug info:
gstbaseaudiosrc.c(583): gst_base_audio_src_create (): /pipeline0/alsasrc0:
dropped 2904 samples
Caught interrupt -- handling interrupt.
Interrupt: Setting pipeline to PAUSED ...
Execution ended after 5372817000 ns.
Setting pipeline to PAUSED ...
Setting pipeline to READY ...
/pipeline0/alsasink0.sink: caps = NULL
/pipeline0/audioresample0.src: caps = NULL
/pipeline0/audioresample0.sink: caps = NULL
/pipeline0/audioconvert0.src: caps = NULL
/pipeline0/audioconvert0.sink: caps = NULL
/pipeline0/alsasrc0.src: caps = NULL
Setting pipeline to NULL ...
FREEING pipeline ...
What can I do to help?
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