Commit graph

54 commits

Author SHA1 Message Date
Doug Nazar
7725c90d5c rtp: Fix request-extension signal call
Signal is registered as taking a guint however was being passed a
guint64 which fails on 32-bit.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1102>
2021-04-28 22:50:53 -04:00
Jakub Adam
538e2ef1d0 rtpbasedepay: fix locking of GstRTPHeaderExtension
'ext' object unlocked if gst_rtp_header_extension_read() fails was never
locked in the first place.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1118>
2021-04-21 17:34:18 +02:00
Jakub Adam
1a87a6572e rtpbasedepayload: handle caps change partway through buffer list
While preparing a blist for pushing, some RTP header extension may
request caps change for a specific buffer in the list. When this
happens, depayloader should immediately push those buffers from the list
that precede the currently processed buffer (for which the caps change
was requested) and only then apply the new caps to the src pad.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1011>
2021-03-12 18:45:04 +01:00
Jakub Adam
c222f322c0 rtphdrext: allow updating depayloader src caps
Add overridable method that updates depayloader's src caps based on
the data from RTP header.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1011>
2021-03-12 18:45:04 +01:00
Guillaume Desmottes
a48edc8372 rtpbasedepayload: add auto-header-extension property
Same property as the one I just added on rtpbasepayload.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1022>
2021-02-03 11:23:40 +01:00
Guillaume Desmottes
df9064fdc6 rtpbasedepayload: set attributes on newly requested extensions
Users were supposed to configure the extension themselves but it was
impossible to do so as they didn't have access to the caps.

Fix #864

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1021>
2021-01-27 09:48:49 +01:00
Guillaume Desmottes
5acde5568e rtpbasedepayload: fix clear-extensions signal definition
Typo as we were using the wrong enum.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1021>
2021-01-27 09:48:49 +01:00
Matthew Waters
092ea647bb rtp/basedepayload: implement support for rtp header extensions
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>
2020-12-03 10:19:32 +00:00
Mikhail Fludkov
d6a2569136 rtpbasedepayload: Mark GAP events sent because of packet loss as such
This allows downstream to distinguish packet loss from normal GAP events
that are sent simply because of gaps in the timeline.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/731>
2020-09-10 16:33:16 +00:00
Havard Graff
5464d420f9 rtpbasedepayload: improve logging around negative gaps
When warning, it is important that the log will contain information to
help debug the problem. Sequence-numbers are crucial here.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/725>
2020-06-26 17:16:33 +00:00
Nicolas Dufresne
8b2afcf56a rtpbasepayload: Save and forward the push flow return
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.
2020-01-11 19:39:55 -05:00
Thibault Saunier
909baa2360 Pass the code through codespell 2019-08-30 13:05:36 +00:00
Mathieu Duponchelle
c854c270be basedepayload: do not create segment in onvif mode
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.
2019-07-18 17:54:04 +02:00
Stian Selnes
eaade96409 rtpbasedepayload: Add max-reorder property
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.
2019-06-13 19:41:11 +03:00
Havard Graff
2e342a16ce rtpbasedepayload: don't consider existing GstRTPSourceMeta
The meta should always be generated based on what is present in the
rtp-header.
2019-06-12 12:38:26 +00:00
Stian Selnes
eadeec791a rtpbasedepayload: Drop gap events before first buffer
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=773104
https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/issues/301
2019-03-20 15:30:50 +00:00
Stian Selnes
f766b85b96 rtpbasepayload: rtpbasedepayload: Add source-info property
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
2018-10-10 14:38:01 -04:00
Tim-Philipp Müller
ca15315565 gst-libs: include config.h in all source files
This will be needed later when we get our export define from config.h
2018-08-13 09:23:34 +01:00
Tim-Philipp Müller
7f9730ecf4 rtp: Update for g_type_class_add_private() deprecation in recent GLib
https://gitlab.gnome.org/GNOME/glib/merge_requests/7
2018-06-23 22:22:22 +02:00
Edward Hervey
924eb8d8a7 rtpbasedepayload: Properly propagate segment seqnum
This wasn't done previously and the outgoing SEGMENT events had
seqnums which weren't consistent with the upstream ones
2018-06-05 17:24:55 +02:00
Mathieu Duponchelle
8467939538 rtpbasedepayload: condition the sending of gap events
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
2018-04-19 16:39:06 +02:00
Thibault Saunier
099ac9faf2 docs: Convert gtkdoc comments to markdown
Modernizing the documentation, making it simpler to read an
modify and allowing us to possibly switch to hotdoc in the
future.
2017-03-10 18:19:17 -03:00
Sebastian Dröge
d84879db28 rtpbasedepayload: Reject non-TIME segments
https://bugzilla.gnome.org/show_bug.cgi?id=765796
2016-11-01 21:09:13 +02:00
Sebastian Dröge
568ec0fc7b Revert "basertpdepayload: create valid segment when given non-time segment"
This reverts commit 0f609bc6c6.
2016-11-01 21:09:04 +02:00
Tim-Philipp Müller
a80c546628 rtp: rtpbasedepayload: simplify code
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.
2016-07-15 19:51:20 +01:00
Zaheer Abbas Merali
0f609bc6c6 basertpdepayload: create valid segment when given non-time segment
This will become an error in 1.10.

https://bugzilla.gnome.org/show_bug.cgi?id=765796
2016-07-01 14:16:46 +02:00
Vivia Nikolaidou
a0cf3b4262 fdmemory, rtpbasedepayload: Ran gst-indent
https://bugzilla.gnome.org/show_bug.cgi?id=764948
2016-04-12 17:34:18 +03:00
Mikhail Fludkov
7a206336dd rtpbasedepayload: look at ssrc before sequence numbers
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
2016-04-03 11:49:16 +03:00
Sebastian Dröge
d6be67265f rtpbasedepayload: Check if the packet loss event actually has timestamp and duration fields
CID 1139615
2015-12-14 13:11:21 +01:00
Tim-Philipp Müller
f0db396e63 rtpbasedepay: when setting discont flag make sure rtpbuffer is current
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.
2015-12-11 11:06:35 +00:00
Tim-Philipp Müller
86350ff8b7 rtpbasedepay: fix possible refcounting issue when detecting a discont
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.
2015-12-11 10:38:14 +00:00
Jan Schmidt
95eb641821 rtpbasedepayload: Make stats creation threadsafe, fix a CRITICAL
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..
2015-08-15 23:37:26 +10:00
Nicolas Dufresne
6ddab6918d basedepayloader: Don't re-timestamp with running-time
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
2015-08-10 13:26:20 -04:00
Tim-Philipp Müller
232bdf1711 rtpbasedepayload: fix leaks in error code paths
This was introduced when reshuffling the buffer unmaps
in commit bc14cdf529
rtp: rtpbasedepayload: add process_rtp_packet() vfunc

Fixes make check-valgrind.

https://bugzilla.gnome.org/show_bug.cgi?id=750235
2015-07-30 12:50:56 +01:00
Nicolas Dufresne
7c638e06ff depayloader: Use input segment start
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
2015-07-18 15:40:26 -04:00
Tim-Philipp Müller
bc14cdf529 rtp: rtpbasedepayload: add process_rtp_packet() vfunc
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
2015-07-12 14:29:29 +01:00
Tim-Philipp Müller
3d7a92b452 rtpbasedepayload: fix typo in comment 2015-07-07 19:56:52 +01:00
Tim-Philipp Müller
fbf2773b2e rtpbasepayload: fix possible segment event leak
Need to clear it when shutting down, not when starting up.
Fixes leak in rtp-payloading unit test.
2015-07-07 15:05:59 +01:00
Tim-Philipp Müller
bc309a100f rtpbasedepayload: provide chain_list function on sink pad
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.
2015-06-01 19:00:55 +01:00
Sebastian Dröge
1a2c590cd2 rtpbasedepayload: Add some debug output 2015-05-05 13:16:05 +02:00
Nicolas Dufresne
b7facbaf22 basedepay: Handle initial gaps and no clock-base
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
2015-03-27 19:03:41 -04:00
Nicolas Dufresne
802ad73103 basedepayload: Fix generated segment
This fixes playback position in RTSP.

https://bugzilla.gnome.org/show_bug.cgi?id=635701
2015-03-26 17:43:47 -04:00
Wim Taymans
f2ee068729 rtpbasedepay: add stats property
Add a stats property that holds a structure with all the current
values of the depayloader.

See https://bugzilla.gnome.org/show_bug.cgi?id=646577
2014-04-12 07:10:36 +02:00
Sebastian Rasmussen
3cc67ff494 rtpbasedepayload: Fix typos in comments 2014-02-24 12:10:26 +01:00
Wim Taymans
121235511a rtpbasedepayload: mark DISCONT on buffer in all cases
Always mark discont on the input buffer when we detect a seqnum
discont and not only when we previously marked ourselves DISCONT.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706422
2013-08-21 12:38:10 +02:00
Tom Greenwood
789ddf42a9 rtpbasedepayload: Ignore caps events if the caps did not change
https://bugzilla.gnome.org/show_bug.cgi?id=697672
2013-04-15 10:00:05 +02:00
Tim-Philipp Müller
5f59b4f7ee Fix FSF address
https://bugzilla.gnome.org/show_bug.cgi?id=687520
2012-11-03 23:05:09 +00:00
Miguel Angel Cabrera Moya
b1318c86e8 rtpbasedepay: remove unused variable
https://bugzilla.gnome.org/show_bug.cgi?id=687146
2012-10-29 21:20:35 +00:00
Mark Nauwelaerts
bd67736851 rtpbasedepay: indicate packet loss using GAP event 2012-09-05 12:02:32 +02:00
Wim Taymans
11a494d5c9 rtp: Add support for multiple memory blocks in RTP
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.
2012-07-17 16:41:36 +02:00