GNOME Bugzilla – Bug 439570
gst not working with my bluetooth headset (Internal GStreamer error: clock problem.)
Last modified: 2007-05-19 13:28:44 UTC
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?
Created attachment 88430 [details] gst-feedback
Try: .... ! queue ! alsasink sync=false (with and without queue), or try with GStreamer core/base CVS which has code to handle pipeline latency.
Work with sync=false (with and without queue)
> Work with sync=false (with and without queue) Great, thanks for confirming. Should be fixed in CVS then (see bug #340842). *** This bug has been marked as a duplicate of 340842 ***