[gstreamer-bugs] [Bug 597463] New: [pulsesrc] has no lower bound for fragment size

[GStreamer (bugzilla.gnome.o. <bug...@gn...>]

https://bugzilla.gnome.org/show_bug.cgi?id=597463
  GStreamer | gst-plugins-good | git

           Summary: [pulsesrc] has no lower bound for fragment size
    Classification: Desktop
           Product: GStreamer
           Version: git
        OS/Version: Linux
            Status: UNCONFIRMED
          Severity: normal
          Priority: Normal
         Component: gst-plugins-good
        AssignedTo: gst...@li...
        ReportedBy: ma...@re...
         QAContact: gst...@li...
      GNOME target: ---
     GNOME version: ---


Since -good commit 1a8938 ("pulsesrc; cleanups, report real latency"), pulsesrc
surrenders control of the segsize to pulseaudio. The latency-time value is
given to pulse as a suggestion, but the actual value returned by PA is blindly
accepted. This can have bad results for applications with higher latency
requirements (intended to reduce buffer handling overhead).

I'm mostly thinking about Maemo 5/N900 here, which has the corner case of a
pulseaudio that is configured to have a 10ms fixed latency source (this will be
fixed in the future). Since this a battery powered device, no application
really wants to actually deal with 10ms audio buffers (not even for VoIP
calls), so we trade in a little latency for some CPU load and battery life.

Attaching proposed patch. The idea is to set the ring buffer segsize to a
larger value if pulseaudio picks a fragsize that is too low. This results in
the ring buffer transparently joining the smaller buffers coming from
pulseaudio into GStreamer buffers of duration equal or close to the requested
latency-time.

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