[gstreamer-bugs] [Bug 641544] New: Baseaudiosink should default to max latency

[GStreamer (bugzilla.gnome.org)]

https://bugzilla.gnome.org/show_bug.cgi?id=641544
  GStreamer | gst-plugins-base | git

           Summary: Baseaudiosink should default to max latency
    Classification: Desktop
           Product: GStreamer
           Version: git
        OS/Version: Linux
            Status: UNCONFIRMED
          Severity: normal
          Priority: Normal
         Component: gst-plugins-base
        AssignedTo: gstreamer-bugs at lists.sourceforge.net
        ReportedBy: bossart.nospam at gmail.com
         QAContact: gstreamer-bugs at lists.sourceforge.net
      GNOME target: ---
     GNOME version: ---


Created an attachment (id=180113)
 --> (https://bugzilla.gnome.org/attachment.cgi?id=180113)
patch

Audio sinks derived from baseaudiosink rely on default buffering and latency
settings, respectively 200 and 10ms. This is not good for media playback (too
many interrupts) and not good for speech (too much latency).
The buffer-time and latency-time can be used to modify the default, but this is
typically not enabled in applications.

Instead of using defaults that are not optimized for any application, the
default should be to use the maximum latency in the audio backend. In case an
application has strict latency requirements (speech calls or gaming) it needs
to set buffer-time and latency-time to relevant values (as done by farsight)
This topic has been addressed a number of times on the mailing lists. See [1]
for example.

You will find attached a set of patches to demonstrate the concept. I added a
new MAX_LATENCY property in the baseaudiosink that is true by default. And I
included alsasink and pulsesink changes to demonstrate how this can be used.

Note that the first patch is really an alsasink bug and should be applied no
matter what the result of this discussion is

-Pierre

[1]
https://tango.0pointer.de/pipermail/pulseaudio-discuss/2010-December/008470.html

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