New signals are added for managing the internal list of rtp header
extension implementations read by a specific depayloader instance.
If the 'extmap-$NUM' field is present in the sink caps, then an
extension implementation will be requested but is not requited to be
able to negotiate correctly. An extension will be requested using the
'request-extension' signal if none could be found internally.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/748>
Save push/push_list helper flow return and in case of failure, return it
in the process function. This allow forwarding downstream flow return
even if the subclass is using the push/push_list helper.
basedepayload generates its own segment in a pretty unconventional
manner, relying on information in the caps such as npt-start or
npt-stop, usually set by rtspsrc.
In ONVIF mode, rtspsrc will generate the correct segment and this
logic in rtpbasedepayload will not be needed, this commit allows
rtspsrc to signal that through the caps.
Add max-reorder property to make the old hard coded reordering limit of
100 configurable. It's particularly useful in some scenarios to set
max-reorder=0 to disable the behavior that the depayloader will drop
packets.
Note that although the default value is 100, the default limit has
increased with one because of the changed if-test. This was done to
allow the max-reorder value to be more intuitive. See tests.
Before a gap event is pushed downstream a segment event must be pushed
since the gap event can cause packet concealment downstream and hence
data flow. Since concealment before receiving any data packets usually
doesn't make any sense, the gap event is not sent downstream.
Alternatively one could generate a default caps and segment event, but
no need to complicate things until it's proven necessary.
https://bugzilla.gnome.org/show_bug.cgi?id=773104https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/issues/301
Add a source-info property that will read/write meta to the buffers
about RTP source information. The GstRTPSourceMeta can be used to
transport information about the origin of a buffer, e.g. the sources
that is included in a mixed audio buffer.
A new function gst_rtp_base_payload_allocate_output_buffer() is added
for payloaders to use to allocate the output RTP buffer with the correct
number of CSRCs according to the meta and fill it.
RTPSourceMeta does not make sense on RTP buffers since the information
is in the RTP header. So the payloader will strip the meta from the
output buffer.
https://bugzilla.gnome.org/show_bug.cgi?id=761947
The default implementation for packet loss handling previously
always sent a gap event.
While this is correct as long as we know the packet that was
lost was actually a media packet, with ULPFEC this becomes
a bit more complicated, as we do not know whether the packet
that was lost was a FEC packet, in which case it is better
to not actually send any gap events in the default implementation.
Some payloaders can be more clever about, for example VP8 can
use the picture-id, and the M and S bits to determine whether
the missing packet was inside an encoded frame or outside,
and thus whether if it was a media packet or a FEC packet,
which is why ulpfecdec still lets these lost events go through,
though stripping them of their seqnum, and appending a new
"might-have-been-fec" field to them.
This is all a bit terrible, but necessary to have ULPFEC
integrate properly with the rest of our RTP stack.
https://bugzilla.gnome.org/show_bug.cgi?id=794909
Remove unnecessary helper struct for callbacks. The bclass
member of the helper struct was not used, so we can just
remove it and the GET_CLASS() call and simplify the whole
affair by passing the depayloader directly to the callback.
Doing so prevents us dropping buffers in the rare, but possible, situations,
when the stream changes SSRC and new sequence numbers does not differ
much from the last sequence number from previous SSRC. For example:
ssrc - 0xaaaa 101,102,103,104 ssrc - 0xbbbb 102, 103, 104, 105...
In the scenario above we don't want to drop the first 3 packets of
0xbbbb stream.
https://bugzilla.gnome.org/show_bug.cgi?id=764459
Depayloaders will look at rtpbuffer->buffer for the discont flag.
When we set the discont flag on a buffer in the rtp base depayloader
and we have to make the buffer writable, make sure the rtpbuffer
actually contains the newly-flagged buffer, not the original input
buffer. This was introduced with the addition of the process_rtp_packet
vfunc, but would only trigger if the input buffer wasn't flagged
already and was not writable already.
When we detect a discont and the input buffer isn't already flagged
as discont, handle_buffer() does a gst_buffer_make_writable() on the
input buffer in order to set the flag. This assumed it had ownership
of the input buffer though, which it didn't. This would still work
fine in most scenarios, but could lead to crashes or mini object
unref criticals in some cases when a discont is detected, e.g. when
using pcapparse in front of a depayloader. This problem was
introduced in bc14cdf529.
Use the object lock to protect the internal segment when updating
against access from getting the stats property.
Fix a critical in gst-inspect or when retrieving the stats
before any segment has arrived by checking whether the
segment has been initted..
There was a confusion, six depayloaders where passing through the
timestamp while the base class was re-timestamping to running
time. This inconstancy has been unnoticed has in most use cases
the incoming segment is [0, inifnity] in which case timestamps are
the same as running time. With DTS/PTS shifting added (to avoid
negative values) and pcapparse sending a different segment this
started being an issue.
https://bugzilla.gnome.org/show_bug.cgi?id=753037
When there is no clock_base provided, the start position is
set to 0 instead of the original segment start value. This
would break synchronization if start was not 0.
https://bugzilla.gnome.org/show_bug.cgi?id=752228
Add process_rtp_packet() vfunc that works just like the
existing process() vfunc only that it takes the GstRTPBuffer
that the base class has already mapped (with MAP_READ),
which means that the subclass doesn't have to map it again,
which allows more performant processing of input buffers
for most RTP depayloaders.
https://bugzilla.gnome.org/show_bug.cgi?id=750235
Implement a chain_list function, which avoids lots of locking
compared to the default fallback implementation in GstPad.
We may also want to do some more sophisticated timestamp
tracking here at some point, but for now leave it up to the
jitterbuffer and/or subclasses (in case buffers in the
buffer list have no timestamp set on them, there may only
be a timestamp for the whole list on the first buffer).
This provides the exact same behaviour as the default
fallback implementation.
When generating segment, we can't assume the first buffer is actually
the first expected one. If it's not, we need to adjust the segment to
start a bit before.
Additionally, we if don't know when the stream is suppose to have
started (no clock-base in caps), it means we need to keep everything in
running time and only rely on jitterbuffer to synchronize.
https://bugzilla.gnome.org/show_bug.cgi?id=635701
Add support RTP buffers with multiple memory blocks. We allow one block for the
header, one for the extension data, N for data and one memory block for the
padding.
Remove the validate function, we validate now when we map because we need to
parse things in order to map multiple memory blocks.
