[gstreamer-bugs] [Bug 439570] New: gst not working with my bluetooth headset

[GStreamer (bugzilla.gnome.org) <bug...@bu...>]

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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: gst...@li...
        ReportedBy: l.b...@ed...
         QAContact: gst...@li...
     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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