GStreamer

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 GStreamer | gstreamer (core) | Ver: 0.10.5

           Summary: "Could not get track start position"
           Product: GStreamer
           Version: 0.10.5
          Platform: Other
        OS/Version: All
            Status: UNCONFIRMED
          Severity: normal
          Priority: Normal
         Component: gstreamer (core)
        AssignedTo: gstreamer-bugs@...
        ReportedBy: joseph.hill@...
         QAContact: gstreamer-bugs@...
     GNOME version: 2.13/2.14
   GNOME milestone: Unspecified


Please describe the problem:
I was trying to rip an oldish CD with sound-juicer and got this output on=
 the
console:

[Thread -1267676240 (LWP 9564) exited]

** (sound-juicer:9552): WARNING **: Could not get track start position

Program received signal SIGSEGV, Segmentation fault.
[Switching to Thread -1225520224 (LWP 9552)]
0xb7f0bf76 in gst_debug_message_get () from /usr/lib/libgstreamer-0.10.so=
.0

I tried ripping the same track with grip and had no problems.

Steps to reproduce:



Actual results:
I get the above output and the track is no ripped.  Sometimes it crashes
sound-juicer, but this is not consistent

Expected results:


Does this happen every time?
Yes, but only with particular tracks.

Other information:


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 GStreamer | gstreamer (core) | Ver: HEAD CVS





------- Comment #3 from Stefan Kost  2006-04-14 17:14 UTC -------
Just to save others of the work - these are the elemnts using the functio=
n that
leaked. I've checked those and IMHO fixing the leak will not break any of
these.

./gst-plugins-bad/ext/faad/gstfaad.c
./gst-plugins-bad/ext/musepack/gstmusepackdec.c
./gst-plugins-bad/ext/swfdec/gstswfdec.c
./gst-plugins-bad/gst/qtdemux/qtdemux.c
./gst-plugins-base/ext/ogg/gstoggdemux.c
./gst-plugins-good/ext/flac/gstflactag.c
./gst-plugins-good/gst/avi/gstavidemux.c
./gst-plugins-good/gst/matroska/matroska-demux.c
./gst-plugins-ugly/ext/dvdnav/dvdnavsrc.c
./gst-plugins-ugly/gst/realmedia/rmdemux.c
./gst-plugins-ugly/gst/asfdemux/gstasfdemux.c


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 GStreamer | gst-plugins-good | Ver: 0.10.x


Wim Taymans changed:

           What    |Removed                     |Added
-------------------------------------------------------------------------=
---
                 CC|                            |wim@...
          Component|don't know                  |gst-plugins-good




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 GStreamer | gst-plugins-base | Ver: HEAD CVS





------- Comment #8 from Wim Taymans  2006-04-14 15:19 UTC -------
disabling sync with the sync=3Dfalse option on the sinks will work around=
 the
problem and will make the sinks follow the rate of the source.


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 GStreamer | gst-plugins-base | Ver: 0.10.5


Wim Taymans changed:

           What    |Removed                     |Added
-------------------------------------------------------------------------=
---
                 CC|                            |wim@...
             Status|UNCONFIRMED                 |RESOLVED
         Resolution|                            |NOTABUG




------- Comment #1 from Wim Taymans  2006-04-14 14:52 UTC -------
buffer-time is the value of the internal audio buffer and should not affe=
ct the
buffer size of the output buffers.

blocksize does nothing for audio sources.

latency-time affects the size of the buffers correctly.

Note that not all sources might be able to adjust their latency-time, in =
the
case of alsa dmix, that value is specified in the alsa configuration file=
 and
cannot be changed by an application.


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 GStreamer | gst-plugins-base | Ver: HEAD CVS





------- Comment #6 from Wim Taymans  2006-04-14 14:17 UTC -------
Created an attachment (id=3D63464)
 --> (http://bugzilla.gnome.org/attachment.cgi?id=3D63464&action=3Dview)
eos after emiting the first infinite duration frame


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 GStreamer | gst-plugins | Ver: 0.8.11


Ronald Bultje changed:

           What    |Removed                     |Added
-------------------------------------------------------------------------=
---
             Status|NEW                         |RESOLVED
         Resolution|                            |FIXED
   Target Milestone|HEAD                        |0.8.13




------- Comment #23 from Ronald Bultje  2006-04-14 13:59 UTC -------
2006-04-14  Frederic Crozat  <fcrozat@...>

        Reviewed by:  Ronald S. Bultje  <rbultje@...>

        * ext/cdparanoia/gstcdparanoia.c: (get_first_sector),
        (cdparanoia_convert):
          Fix no-track-mode (e.g. gnome-cd), while not breaking non-track
          mode applications (e.g. Goobox). Fixes #318574.


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 GStreamer | gst-plugins-base | Ver: HEAD CVS





------- Comment #7 from Wim Taymans  2006-04-14 13:35 UTC -------
Created an attachment (id=3D63461)
 --> (http://bugzilla.gnome.org/attachment.cgi?id=3D63461&action=3Dview)
patch to reenable more audio formats

adding the other supported formats to the caps and activating them seems =
to
work fine for me.


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 GStreamer | gst-plugins-base | Ver: 0.8.11





------- Comment #6 from F. Steinel  2006-04-14 09:42 UTC -------
for hints see:
http://svn.beep-media-player.org/bmpx/trunk/src/libvisual.c
and
http://cvs.sourceforge.net/viewcvs.py/libvisual/libvisual-hackground/stan=
dalone/


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 GStreamer | gst-plugins-base | Ver: HEAD CVS

           Summary: Regression in the handling of files with multiple
                    audio/subtitle streams
           Product: GStreamer
           Version: HEAD CVS
          Platform: Other
        OS/Version: Linux
            Status: UNCONFIRMED
          Severity: normal
          Priority: Normal
         Component: gst-plugins-base
        AssignedTo: gstreamer-bugs@...
        ReportedBy: thaytan@...
         QAContact: gstreamer-bugs@...
     GNOME version: Unspecified
   GNOME milestone: Unspecified


I just discovered a file that refuses to preroll properly in playbin. Aft=
er
investigating, it turns out that the file has 2 audio tracks.=20

Decodebin relies on pad_probes to detect data passing each decoder source=
 pad.
In this case, the 2nd audio track never triggers the pad probe because it=
 never
outputs a buffer. Instead, it is just performing a buffer-alloc and disco=
vering
that downstream is not linked (It is not the selected track and the
StreamSelector is returning not-linked)

There are 2 solutions:
1) Get decodebin to also detect an attempt at buffer-allocation and mark =
that
pad as prerolled. pad_probes do not get called for buffer-alloc, so this =
will
either need core changes, or decodebin needs changing to block the pads
instead.
2) Restore the old behaviour of StreamSelector to always perform a fallba=
ck
buffer-allocation, even for non-selected pads.=20

Since 2) is less intrusive, and fixes the regression, attaching a patch f=
or
that.


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 GStreamer | gstreamer (core) | Ver: HEAD CVS





------- Comment #6 from Jan Schmidt  2006-04-14 04:12 UTC -------
Then there ain't nuthin' to it but to do it


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