mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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f52e16ceb8
This reverts commit dcd3ce9751.
This functionality was implemented for gstopenwebrtc, but it
turned out this was not actually needed for webrtc bundling
support, as shown in webrtcbin. It also doesn't correspond
to any standards.
This is an API break, but nothing should actually depend on
this, at least not for its initial purpose.
Changes in rtpbin.c were reverted manually, to preserve some
refactoring that had occurred in the original commit.
Fixes #537
86 lines
906 B
Text
86 lines
906 B
Text
.dirstamp
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aacparse
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ac3parse
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alpha
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alphacolor
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amrparse
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apev2mux
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aspectratiocrop
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audioamplify
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audiochebband
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audiocheblimit
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audioecho
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audiowsincband
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audiowsinclimit
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audiodynamic
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audiofirfilter
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audioinvert
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audioiirfilter
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audiopanorama
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autodetect
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avimux
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avisubtitle
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capssetter
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deinterlace
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deinterleave
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dtmf
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equalizer
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gdkpixbufoverlay
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gdkpixbufsink
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flacparse
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flvdemux
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flvmux
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icydemux
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id3demux
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id3v2mux
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imagefreeze
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interleave
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jpegdec
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jpegenc
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level
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matroskademux
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matroskamux
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matroskaparse
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mpegaudioparse
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mpg123audiodec
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mulawdec
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mulawenc
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multifile
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qtdemux
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qtmux
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rganalysis
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rglimiter
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rgvolume
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rtp-payloading
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rtpbin
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rtpbin_buffer_list
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rtpcollision
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rtph261
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rtph263
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rtph264
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rtpjitterbuffer
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rtpred
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rtpsession
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rtpstorage
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rtpmux
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rtprtx
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rtpulpfec
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rtpvp9
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shapewipe
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souphttpsrc
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spectrum
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splitmux
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udpsink
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udpsrc
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videocrop
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videobox
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videofilter
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videomixer
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vp8dec
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vp8enc
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vp9enc
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wavpackdec
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wavparse
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wavpackenc
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wavpackparse
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y4menc
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