So far we assumed that if all sources are bye, this meant we needed to
send an EOS on the RTCP sink. The problem is that this case may happens
if we only had one internal source and it detected a collision.
So now we limit the EOS forwarding to when there is a send_rtp_sink pad
and that this pad has received EOS. We don'tcheck the recv_rtp_sink
since the code does not wait for the bye to be send before sending EOS
to the RTCP src pad.
RF64 encode support was added to wavenc quite some time
ago, but not declared in wavparse. It seems wavparse can
decode it though, so add it to the sink pad.
The RF64 support was added in
https://bugzilla.gnome.org/show_bug.cgi?id=735627
This is useful when implementing custom retransmission mechanism like
RIST to prevent RTCP from being produces for the retransmitted SSRC.
This would also be used in general for various purpose when customizing
an RTP base pipeline.
If it goes over 2^15 packets, it will think it has rolled over
and start dropping all packets. So make sure the seqnum distance is not too big.
But let's not limit it to a number that is too small to avoid emptying it
needlessly if there is a spurious huge sequence number, let's allow at
least 10k packets in any case.
Recent changes in ccextractor were attaching timecode meta to the closed
caption track. We shouldn't write timecode information for the closed
caption trak.
It is possible that PulseAudio adds formats that are not yet supported
in pulsesink, and in those cases, we want to gracefully skip them rather
than cause an assert on a NULL caps.
And let it the oportunity to get its other pad linked
Example:
```
$ gst-launch-1.0 uridecodebin uri=file:///home/thiblahute/gst-validate.save/gst-integration-testsuites/testsuites/../medias/defaults/flv/819290236.flv caps=audio/x-raw expose-all-streams=FALSE ! fakesink
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
ERROR: from element /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0/GstDecodeBin:decodebin0/GstFlvDemux:flvdemux0: Internal data stream error.
Additional debug info:
../subprojects/gst-plugins-good/gst/flv/gstflvdemux.c(2760): gst_flv_demux_loop (): /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0/GstDecodeBin:decodebin0/GstFlvDemux:flvdemux0:
streaming stopped, reason not-linked (-1)
ERROR: pipeline doesn't want to preroll.
Setting pipeline to NULL ...
Freeing pipeline ...
```
Modify the caps string to allow width and height greater than 4096.
There is no need to restrict it since the matroska format allows the
width and height values to be up to eight bytes long, and this also
applies to the webm subset of the format.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/550
This was originally added for fixing conflicting definitions between
Android and Qt, but times have changed and now this breaks the build
on iOS:
[...]/OpenGLES.framework/Headers/ES3/gl.h:1006:48: error: unknown type name 'gst_qt_GLsync'
GL_API void GL_APIENTRY glGetSynciv (GLsync sync, GLenum pname, GLsizei bufSize, GLsizei* length, GLint* values) OPENGLES_DEPRECATED(ios(7.0, 12.0), tvos(9.0, 12.0));
^
../ext/qt/gstqtgl.h:49:16: note: expanded from macro 'GLsync'
#define GLsync gst_qt_GLsync
^
6 errors generated.
Instead, we simply avoid defining GLsync ourselves if we're using Qt.
When multiple nals are aggrgated, the marker bit should be associated only
with the last NAL of the packet. Otherwise we may break rendering in with
AU alignment.
Allow fallback to orc subproject if any.
Additionally 'dependencies' keyword is removed from find_library,
because it's invalid keyword for find_library.
It makes no sense for a decoder to output a BYTES segment, and
many elements one would plug downstream of a video decoder assume
the segments they receive are in TIME format, for example this fixes:
gst-validate-1.0 filesrc location=opacity01.svg ! gdkpixbufdec ! \
videobalance ! videoconvert ! fakesink
In that case, videobalance was emitting an assertion when trying to
call gst_object_sync_values()
