If the id of a #FsCodec is #FS_CODEC_ID_ANY, then it will be replaced with a dynamic payload type at runtime
If the id of a #FsCodec is #FS_CODEC_ID_DISABLE, then this codec will not be used
A format that can be used in printf like format strings to format a FsRtpHeaderExtension
Reads the content of a #GKeyFile of the following format into a #GList of #FsCodec structures.
Example: |[ [audio/codec1] clock-rate=8000
[audio/codec1:1] clock-rate=16000
[audio/codec2] one_param=QCIF another_param=WOW
[video/codec3] wierd_param=42 feedback:nack/pli=1 feedback:tfrc=
@param filename Name of the #GKeyFile to read the codecs parameters from
Does a deep copy of a #GList of #FsRtpHeaderExtension
a #GList of #FsRtpHeaderExtension
Reads the content of a #GKeyFile of the following format into a #GList of #FsRtpHeaderExtension structures.
The groups have a format "rtp-hdrext:audio:XXX" or "rtp-hdrext:video:XXX" where XXX is a unique string (per media type).
The valid keys are:
Example: |[ [rtp-hdrext:audio:a] id=1 uri=urn:ietf:params:rtp-hdrext:toffset
[rtp-hdrext:audio:abc] id=3 uri=urn:ietf:params:rtp-hdrext:ntp-64 direction=receive
@param filename Name of the #GKeyFile to read the RTP Header Extensions from
@param mediaType The media type for which to get header extensions
These default rtp header extension preferences should work with the elements that are available in the main GStreamer element repositories. They should be suitable for standards based protocols like SIP or XMPP.
Element for which to fetch default RTP Header Extension preferences
The #FsMediaType for which to get default RTP Header Extension preferences
This allows setting the bitrate on all elements that have a "bitrate" property without having to know the type or of the unit used by that element.
This will be obsolete in 0.11 (when all elements use bit/sec for the "bitrate" property.
A format that can be used in printf like format strings to format a FsCodec