[gstreamer-bugs] [Bug 613248] Interop: no mic when calling Polycom Soundpoint phone.

Thu Apr 8 10:05:30 PDT 2010

https://bugzilla.gnome.org/show_bug.cgi?id=613248
  GStreamer | gst-plugins-base | git

--- Comment #4 from Olivier Crete (Tester) <tester at tester.ca> 2010-04-08 17:05:25 UTC ---
Created an attachment (id=158220)
 View: https://bugzilla.gnome.org/attachment.cgi?id=158220
 Review: https://bugzilla.gnome.org/review?bug=613248&attachment=158220

Make it possible to force buffers that are a multiple of X ns

Here's a patch that adds a new property to GstBaseRtpAudioPayload to force
output buffers to be an exact multiple of a specified duration. Then we can
force it to 10ms when calling SIP phones and our lives will be happier!

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