[Bug 641072] [baseaudiosink] Improve latency-time handling and optionally aggregate buffers until latency-time is reached

Mon Apr 11 08:58:11 PDT 2011

https://bugzilla.gnome.org/show_bug.cgi?id=641072
  GStreamer | gst-plugins-base | 0.10.x

Pierre Bossart <bossart.nospam> changed:

           What    |Removed                     |Added
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                 CC|                            |bossart.nospam at gmail.com

--- Comment #8 from Pierre Bossart <bossart.nospam at gmail.com> 2011-04-11 15:58:07 UTC ---
FYI, this bug is related to the changes I suggested in #641544.
I think we all agree that the default buffer sizes aren't great for music
playback and could be increased.
However as Arun pointed out, increasing buffer sizes up to the default
pulseaudio 2s has pretty nasty side effects on existing applications. Totem
takes forever to pause and Rhythmbox ends-up crapping out... Instead of forcing
a large buffer in playbin2 we may want the application to use a profile to
indirectly set theses values.

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