[gstreamer-bugs] [Bug 315121] [alsasink] doesn't work unless passing hw:0

GStreamer (bugzilla.gnome.org) bugzilla-daemon at bugzilla.gnome.org
Mon Oct 31 03:19:19 PST 2005


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http://bugzilla.gnome.org/show_bug.cgi?id=315121
 GStreamer | gst-plugins | Ver: 0.8.10

Tim-Philipp Müller changed:

           What    |Removed                     |Added
----------------------------------------------------------------------------
  Attachment #53329|0                           |1
        is obsolete|                            |
            Summary|alsasink doesn't work unless|[alsasink] doesn't work
                   |passing hw:0                |unless passing hw:0



------- Additional Comments From Tim-Philipp Müller  2005-10-31 11:19 -------
Okay, I've committed a modified version of the patch above, so that the extra
probing can be switched off using the 'advanced-probing' property.

2005-10-31  Tim-Philipp Muller  <tim at centricular dot net>

        * ext/alsa/gstalsa.c: (gst_alsa_class_init),
        (gst_alsa_set_property), (gst_alsa_get_property),
        (gst_alsa_get_caps), (gst_alsa_open_audio_device),
        (gst_snd_pcm_info_get_real_device), (gst_alsa_open_audio):
        * ext/alsa/gstalsa.h:
          When the default device is being used, try to probe the caps
          of the underlying device instead if possible. This should give
          us more narrowly defined caps that are closer to the hardware's
          capabilities. This is enabled by default, but can be switched
          off via the new 'advanced-probing' property (#315121).


I hope that this, in combination with the fixes from bug #318767 and bug
#318273, should solve a lot of problems people have been having with the
GStreamer alsasink.


Some of the above debian.org bugs are not GStreamer bugs though, but due to
broken pipelines. When decoding ogg/vorbis files, the result with be float
audio. audioscale cannot process float audio, so you get that 'failed to set
caps' error. It should work though if you put an 'audioconvert' right after spider.


Note also that the patch will give more narrowly defined caps, which means that
having audioconvert/audioscale in the pipeline (in the right order) is even more
crucial than it was before. For example, on my debian sid dmix is used by
default, so with the above patch the caps accepted will only be

  audio/x-raw-int, width=(int)16, depth=(int)16, signed=(boolean)true,
endianness=(int)1234, rate=(int)48000, channels=(int)2

so even something like

  gst-launch-0.8 sinesrc ! alsasink

won't work anymore without at least an audioconvert in the middle (to convert
from the 1 channel sinesrc produces to the 2 channels alsasink wants).

This might be a bit of a hassle in the short term, but any applications or
pipelines that don't work after the changes use broken pipelines anyway and need
to be fixed.



Please give these changes a spin and let me know how it goes.

Cheers
 -Tim


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