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
The initial mission statement for this test was:
* demonstrate usage of the request-aux-* signals in rtpbin
* test the rtx elements
We have examples that serve the first use case, and better
(harnessed) tests for the second use case.
This test is slow and racy, it served its purpose but can now
be removed.
Fixes#533
Even though hooked up to the build system, it's clear that no one
has ever built or used this with GStreamer 1.x. It wants to link
against libgstinterfaces, which no longer exists. And uses 0.10-style
raw audio caps. And the last meaningful change was done in 2009.
Let's just remove it.
A new signal named on-bundled-ssrc is provided and can be
used by the application to redirect a stream to a different
GstRtpSession or to keep the RTX stream grouped within the
GstRtpSession of the same media type.
https://bugzilla.gnome.org/show_bug.cgi?id=772740
Some of the subtitle chunks will have embedded
NUL-terminators (last three), some don't (first three),
some will have markup, some won't, some will be valid
UTF-8 (all but last), some won't (last stanza).
https://bugzilla.gnome.org/show_bug.cgi?id=752421
Implement 2 new elements - splitmuxsink and splitmuxsrc.
splitmuxsink is a bin which wraps a muxer and takes 1 video stream,
plus audio/subtitle streams, and starts a new file
whenever necessary to avoid overrunning a threshold of either bytes
or time. New files are started at a keyframe, and corresponding audio
and subtitle streams are split at packet boundaries to match
video GOP timestamps.
splitmuxsrc is a corresponding source element which handles
the splitmux:// URL and plays back all component files,
reconstructing the original elementary streams as it goes.
It shows how to use "set-aux-receive" and "set-aux-send"
properties of rtpbin to set rtprtxsend and rtprtxreceive
Build 2 pipelines, one for rtpbin as a sender and one for
rtobin as a receive. Then transmit an audio stream.
It also drops some packets to activate restransmission and
check they are actually retransmited.
This never really took off and is most likely completely
unused. If there is still a need for this, it should
probably be done differently, perhaps inside oggdemux/mux.
Original commit message from CVS:
* docs/plugins/.cvsignore:
* tests/check/elements/.cvsignore:
Ignore some more generated things
* tests/check/Makefile.am:
Ignore OSS elements in the state changes test too.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/gdkpixbufsink.c:
Add unit test for gdkpixbufsink element.
