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Bug 340909 - [alsasink] can't enable spdif output
[alsasink] can't enable spdif output
Status: RESOLVED FIXED
Product: GStreamer
Classification: Platform
Component: gst-plugins-base
git master
Other All
: Normal major
: 0.10.8
Assigned To: GStreamer Maintainers
GStreamer Maintainers
Depends on:
Blocks:
 
 
Reported: 2006-05-07 09:42 UTC by Christoph Burghardt
Modified: 2006-05-23 14:33 UTC
See Also:
GNOME target: ---
GNOME version: 2.15/2.16



Description Christoph Burghardt 2006-05-07 09:42:12 UTC
Please describe the problem:
ela:~$ LANG=C gst-launch audiotestsrc wave=1 ! alsasink
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
Pipeline is PREROLLED ...
Setting pipeline to PLAYING ...
New clock: audioclock0

Works fine. If i select the default device explicit:

ela:~$ LANG=C gst-launch audiotestsrc wave=1 ! alsasink device=hw:0,0
Setting pipeline to PAUSED ...
ERROR: Pipeline doesn't want to pause.
ERROR: from element /pipeline0/audiotestsrc0: Could not negotiate format
Additional debug info:
gstbasesrc.c(1717): gst_base_src_start (): /pipeline0/audiotestsrc0:
Check your filtered caps, if any
Setting pipeline to NULL ...
FREEING pipeline ...

Does not :(  So i expand the pipeline to

ela:~$ LANG=C gst-launch audiotestsrc wave=1 ! audioconvert ! audiorate !
alsasink device=hw:0,0
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
Pipeline is PREROLLED ...
Setting pipeline to PLAYING ...
New clock: audioclock0

which works fine again, but if i now change to the spdif output (hw:0,4)

ela:~$ LANG=C gst-launch audiotestsrc wave=1 ! audioconvert ! audiorate !
alsasink device=hw:0,4
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
ERROR: from element /pipeline0/alsasink0: Could not get/set settings from/on
resource.
Additional debug info:
gstalsasink.c(604): set_hwparams (): /pipeline0/alsasink0:
Rate doesn't match (requested 44100Hz, get 0Hz)
ERROR: pipeline doesn't want to preroll.
Setting pipeline to NULL ...
FREEING pipeline ...

A simple 

gst-launch sinesrc ! alsasink device=hw:0,4 works on this computer with gst-0.8

Steps to reproduce:


Actual results:


Expected results:


Does this happen every time?


Other information:
Comment 1 Michael Smith 2006-05-11 20:22:08 UTC
Couple of problems here (discussed a bit on irc):

 - should be using audioresample, not audiorate.
 - with that, it works if we use filtered caps to specify that audioresample should resample to 48000. Otherwise, it still fails.

Guess this is a problem with probing capabilities in alsasink?
Comment 2 Tim-Philipp Müller 2006-05-18 13:04:43 UTC
Sample rate probing should work in gst-plugins-base CVS now:


  2006-05-15  Tim-Philipp Müller  <tim at centricular dot net>

        * ext/alsa/gstalsasink.c: (gst_alsasink_getcaps):
          Query supported sample rates. Fixes #341732.


Can this be closed or is there something else involved?
Comment 3 Tim-Philipp Müller 2006-05-23 14:33:21 UTC
Closing, should be fixed now.

It should be

    audiotestsrc ! audioconvert ! audioresample ! alsasink


btw (you need the audioconvert to convert between formats and from mono <=> stereo and the audiorsample to convert between sample rates).

Please re-open if this is still an issue with -base CVS.