gst_buffer_map () results in memcopying when a GstBuffer contains
more than one GstMemory and when AVC (length-prefixed) alignment is used.
This has quite an impact on performance on systems with limited amount of
resources. With this patch the whole GstBuffer will not be mapped at once,
instead each individual GstMemory will be iterated and mapped separately.
Add parsed=true to output caps, as we always output
whole frames, timestamped and all. Means also that
the output can be decoded by avdec_mjpeg wihout
plugging an extra parser (which has no rank).
By passing `NULL` to `g_signal_new` instead of a marshaller, GLib will
actually internally optimize the signal (if the marshaller is available
in GLib itself) by also setting the valist marshaller. This makes the
signal emission a bit more performant than the regular marshalling,
which still needs to box into `GValue` and call libffi in case of a
generic marshaller.
Note that for custom marshallers, one would use
`g_signal_set_va_marshaller()` with the valist marshaller instead.
If VP8 is not encoded with error resilience enabled then any packet loss
causes very bad artefacts when decoding and waiting for the next
keyframe instead improves user experience considerably.
adjust/port from rtph264pay and allow sending the configuration data at
every IDR
The payloader was stripping the configuration data when the
config-interval was set to 0. The code was written in such a way !(a >
0) that it stripped the config when it was set at -1 (send config_data
as soon as possible).
This resulted in some MPEG4 streams where no GOP/VOP-I was detected to
be sent out without configuration.
Add a new property "do-aggregate"* to the H.264 RTP payloader which
enables STAP-A aggregation as per [RFC-6184][1]. With aggregation enabled,
packets are bundled instead of sent immediately, up until the MTU size.
Bundles also end at access unit boundaries or when packets have to be
fragmented.
*: The property-name is kept generic since it might apply more widely,
e.g. STAP-B or MTAP.
[1]: https://tools.ietf.org/html/rfc6184#section-5.7
Closes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/434
Otherwise it can happen that we receive a caps event, then another caps
event and only then buffers. We would then send out the first caps event
in the stream but mark buffers with the caps version of the second caps
event.
Otherwise it can happen that we already collected 7 caps, miss the 8th
caps packet (packet loss) and then re-use the 1st caps for the following
buffers instead of the 8th caps which will likely cause errors further
downstream unless both caps are accidentally the same.
Keeping old caps around does not seem to have any value other than
potentially causing errors. We would always receive new caps whenever
they change (even if they were previous ones) and it's very unlikely
that they happen to be exactly the same as the previous ones.
Also after having received new caps or a buffer with a next caps
version, no buffers with old caps version will arrive anymore.
Fix doc chunks to not use that syntax for links that have the
url as description, it will be put verbatim into the xml/*.xml
file and then the expat parser will throw a syntax error like:
File "../../common/mangle-db.py", line 71, in <module>
main()
File "../../common/mangle-db.py", line 69, in main
patch (details.replace("-details", ""), os.path.basename(details))
File "../../common/mangle-db.py", line 20, in patch
doc = xml.dom.minidom.parse(related)
File "/usr/lib/python2.7/xml/dom/minidom.py", line 1918, in parse
return expatbuilder.parse(file)
File "/usr/lib/python2.7/xml/dom/expatbuilder.py", line 924, in parse
result = builder.parseFile(fp)
File "/usr/lib/python2.7/xml/dom/expatbuilder.py", line 207, in parseFile
parser.Parse(buffer, 0)
xml.parsers.expat.ExpatError: not well-formed (invalid token): line 84, column 7
If the incoming frame buffer has GST_BUFFER_FLAG_DISCONT set this should
be preserved and set for the first output buffer too, like other
payloaders do.
Spotted with gst-validate-1.0 when adding integration tests for
rtpsession, a minimal test to reproduce the issue is:
$ gst-validate-1.0 videotestsrc num-buffers=1 ! rtpvrawpay ! identity ! fakesink
Starting pipeline
Pipeline started
warning : Buffer didn't have expected DISCONT flag333 speed: 1.000000 />
Detected on <identity0:sink>
Detected on <identity0:src>
Detected on <fakesink0:sink>
Description : Buffers after SEGMENT and FLUSH must have a DISCONT flag
Issues found: 1
=======> Test PASSED (Return value: 0)
When used in combination with a rtponviftimestamp element
downstream, forwarding this flag ensures it gets correctly
serialized in the ONVIF header extension.
A missing colon after G_DEFINE_TYPE declaration was confusing gst-indent
and causing problem in the pre-commit hook.
Add the missing colon and fix the following function declaration to
follow the normal GStreamer style.
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.
This macro is not longer used. It was secretly checking if that nal was
a slice, and confusingly name to that one may think it was checking if
the nal is an AUD.
The code was reading the timestamp from the adapter before pushing the
new buffer into it. As a side effect, if the adapter was empty, we'd end
up using an older timestamp. In alignment=au, it means that all
timestamp was likely one frame in the past, while in alignment=nal, with
multiple slices per frame, the first slice would have the timestamp of
the previous one.
The marker bit is used for efficient decoding. The assumption that
it should be set on the AUD is wrong, since the AUD is conceptually
starts the frame, while the marker is to indicate the end.
So properly set the marker bit as soon as we know we are ending an
AU and also whenever upstream have set the GST_BUFFER_FLAG_MARKER
flag.
The code was reading the timestamp from the adapter before pushing the
new buffer into it. As a side effect, if the adapter was empty, we'd end
up using an older timestamp. In alignment=au, it means that all
timestamp was likely one frame in the past, while in alignment=nal, with
multiple slices per frame, the first slice would have the timestamp of
the previous one.
The marker bit is used for efficient decoding. The assumption that
it should be set on the AUD is wrong, since the AUD is conceptually
starts the frame, while the marker is to indicate the end.
So properly set the marker bit as soon as we know we are ending an
AU and also whenever upstream have set the GST_BUFFER_FLAG_MARKER
flag.
Don't allow external encoder to use one of the reserved NAL type
implicated in NAL aggreation. These out-of-spec NAL types, if passed
from the outside world will lead to an invalid RTP payload being
created.
Strip ADTS headers if we detect any, apparently some Sony cameras
send AAC with ADTS headers. We could also change the stream-format
in the output caps, but that would be unexpected to pipeline builders
and would not exactly be backwards compatible.
Both rtpmp4vpay and rtpmp4gpay support MPEG4 elementary streams. But
the most supported variant is the video-specific one (rtpmp4vpay),
therefore increase the rank of that one so that auto-plugging of
payloaders for MPEG4 elementary streams ends up picking that one
and not the generic one.
rtph264pay and rtph265pay skip updating the parameter set timestamp if
the units they see contain no new configuration. This can result in
them injecting duplicate parameters.
https://bugzilla.gnome.org/show_bug.cgi?id=796748
All these were copy pasted and would lead to assertion when chained with
rtpmux. This commit rewrite the negotiation with downstream. This also
drop the fallback to ancient names if the pad is unlinked. This was
completly arbitrary decision that made no sense.
https://bugzilla.gnome.org/show_bug.cgi?id=796809
Just remove the code. It's not doing anything useful anyways. The modified
caps are the result of a caps query, so either not used afterwards of a
reference to some internal caps of another element that should not be
modified.
https://bugzilla.gnome.org/show_bug.cgi?id=796837
ULP FEC, as defined in RFC 5109, has the protected and protection
packets sharing the same ssrc, and a different payload type, and
implies rewriting the seqnums of the protected stream when encoding
the protection packets. This has the unfortunate drawback of not
being able to tell whether a lost packet was a protection packet.
rtpbasedepayload relies on gaps in the seqnums to set the DISCONT
flag on buffers it outputs. Before that commit, this created two
problems:
* The protection packets don't make it as far as the depayloader,
which means it will mark buffers as DISCONT every time the previous
packets were protected
* While we could work around the previous issue by looking at
the protection packets ignored and dropped in rtpptdemux, we
would still mark buffers as DISCONT when a FEC packet was lost,
as we cannot know that it was indeed a FEC packet, even though
this should have no impact on the decoding of the stream
With this commit, we consider that when using ULPFEC, gaps in
the seqnums are not a reliable indicator of whether buffers should
be marked as DISCONT or not, and thus rewrite the seqnums on
the decoding side as well to form a perfect sequence, this
obviously doesn't prevent the jitterbuffer from doing its job
as the ulpfec decoder is downstream from it.
https://bugzilla.gnome.org/show_bug.cgi?id=794909
Fix compilation with MSVC. We still assume that attribute
is supported by all other relevant compilers, which seems
to be the case since we haven't had any complaints about
similar code in rtpsbcpay.
