[gstreamer-bugs] [Bug 524724] New: [PATCH] [baseaudiosrc] buffer-time and latency-time do not reflect actual values

[GStreamer (bugzilla.gnome.org)]

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  GStreamer | gst-plugins-base | Ver: 0.10.18
           Summary: [PATCH] [baseaudiosrc] buffer-time and latency-time do
                    not reflect actual values
           Product: GStreamer
           Version: 0.10.18
          Platform: Other
        OS/Version: Linux
            Status: UNCONFIRMED
          Severity: normal
          Priority: Normal
         Component: gst-plugins-base
        AssignedTo: gstreamer-bugs at lists.sourceforge.net
        ReportedBy: manauw at skynet.be
         QAContact: gstreamer-bugs at lists.sourceforge.net
     GNOME version: Unspecified
   GNOME milestone: Unspecified


That is to say, the application/user can set buffer-time and latency-time (more
or less) as desired.  Later on, in gst_base_audio_src_setcaps, ringbuffer specs
are initialized with these values, and eventually (typically) passed onto an
element's _prepare (e.g. alsasrc, osssrc) which then (possibly heavily) adjusts
these specs.  Upon return to _set_caps, the ringbuffer's buffer-time and
latency-time are recalculated to reflect these changes, but this is not
propagated to the element-properties latency-time and buffer-time.

This may very well be as intended (?), but it does leave these properties with
values that have very little (if any) "real meaning".  In particular, the
"outside world" (application/user) is left with no clue as to what is actually
happening, i.e. being used.


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