Commit graph

501 commits

Author SHA1 Message Date
Sebastian Dröge
9156b373e6 rtpbin: Regularly emit the sync signal
Even if no new synchronization information is available.

This is necessary because the timestamp offset logic in rtpbin depends
on the base RTP time that is determined by the jitterbuffer, but this
changes all the time (especially in mode=slave) and the timestamp
offsets have to be updated accordingly. Doing so is especially important
if they're only determined by the RTP-Info, which never changes from the
very beginning.

The interval can be configured via the new min-sync-interval property.
Synchronization happens at least that often, but at most as often as the
old sync-interval property allows.
Both intervals are now based on the monotonic system clock.

Additionally, clean up synchronization code a bit, only emit either
inband NTP or RTCP SR synchronization at the same time, based on which
one has the more recent time information, and only emit RTP-Info
synchronization if it wasn't provided previously at the same time as the
NTP-based synchronization information.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6543>
2024-05-28 11:52:31 +00:00
Sebastian Dröge
df8c29e340 rtpjitterbuffer: Set max-rtcp-rtp-sync-time to -1 (disabled)
There is generally no requirement to ignore RTCP SR if the RTP time of
the SR differs a lot from the last received RTP packet. The mapping
between RTP and NTP time stays valid until there was a stream reset, in
which case we wouldn't use that information anyway.

When using rtcp-sync-send-time=false the default of 1s difference can
easily be exceeded, e.g. if encoding of the stream after capture adds
more than 1s of latency.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6543>
2024-05-28 11:52:30 +00:00
Sebastian Dröge
95a0649945 rtpbin: Allow synchronizing against RTP-Info without having received any RTCP
Previously the information was provided from rtpjitterbuffer to rtpbin
only once the first RTCP SR was received, which is not necessary at all
as all required information is available from the caps already.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1162

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6543>
2024-05-28 11:52:30 +00:00
Sebastian Dröge
8bfba72ea4 rtpbin: Add new never/ntp RTCP sync modes
Never is useful for some RTSP servers that report plain garbage both via
RTCP SR and RTP-Info, for example.

NTP is useful if synchronization should only ever happen based on RTCP
SR or NTP-64 RTP header extension.

Also slightly change the behaviour of always/initial to take RTP-Info
based synchronization into account too. It's supposed to give the same
values as the RTCP SR and is available earlier, so will generally cause
fewer synchronization glitches if it's made use of.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6543>
2024-05-28 11:52:30 +00:00
Sebastian Dröge
158f12b5da rtpbin: Handle switches between RTP-Info and NTP-based stream association better
Instead of switching on the very first stream, require that all streams
have switched before switching to the different synchronization
mechanism.

Without this there will be a noticeable gap during the switch. E.g. when
going from RTP-Info to NTP-based association, first the first stream
only would get an offset, then the first two, ... then all of them.
Depending on the order of streams this will cause a lot of changes in
ts-offset during the transition.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6543>
2024-05-28 11:52:30 +00:00
Sebastian Dröge
b30671a8ee rtpbin: Pass NPT start from rtpjitterbuffer to rtpbin
And use it to detect synchronization changes (e.g. seeks) more reliably
when doing RTP-Info based synchronization.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6543>
2024-05-28 11:52:30 +00:00
Sebastian Dröge
3eb22af88b rtpbin: Clean up stream association state
Use fewer magic numbers and keep track of the different synchronization
mechanisms separately. Also keep track of more state to detect more
situations when resynchronization should happen.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6543>
2024-05-28 11:52:30 +00:00
Sebastian Dröge
d8dabf142f rtpbin: Constify function parameters and use correct types
Previously these parameters were randomly changed in the body of the
function to avoid having to declare a new variable, which made the code
very hard to follow. By marking them as const this won't be possible
anymore in the future.

Also the RTP clock-base (RTP time from RTSP RTP-Info) is an unsigned
64 bit integer as it's an extended RTP timestamp.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6543>
2024-05-28 11:52:30 +00:00
Sebastian Dröge
155c3fb3b2 rtpbin: Untangle NTP-based and RTP-Info based stream association
Both were entangled previously and very hard to follow what happens
under which conditions. Now as a very first step the code decides which
of the two cases it is going to apply, and then proceeds accordingly.
This also avoids calculating completely invalid values along the way and
even printing them int the debug output.

Also improve debug output in various places.

This shouldn't cause any behaviour changes.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6543>
2024-05-28 11:52:30 +00:00
Sebastian Dröge
7d0c7144ba rtpbin: Remove unused variable / function parameter
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6543>
2024-05-28 11:52:30 +00:00
Sebastian Dröge
4421c3de75 rtpbin: Handle ntp-sync=true before everything else
This simplifies the code as it's a much simpler case than the normal
inter-stream synchronization, and interleaving it with that only
reduces readability of the code.

Also improve some debug output in this code path.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6543>
2024-05-28 11:52:30 +00:00
Sebastian Dröge
4b0e75a094 rtpbin: Add some documentation to gst_rtp_bin_associate()
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6543>
2024-05-28 11:52:30 +00:00
Sebastian Dröge
70a435c0c4 rtpbin: Don't do any timestamp offsetting in rfc7273-sync=true mode
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1160

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6543>
2024-05-28 11:52:30 +00:00
Sergey Krivohatskiy
1c5e1798b6 flacparse: fix buffer overflow in gst_flac_parse_frame_is_valid
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6835>
2024-05-27 23:31:44 +00:00
Tim-Philipp Müller
8bd1a3213e level: fix old "message" property doc chunk
In the online documentation the new post-messages
property would show up as deprecated refering to
itself.

Fixes #3561

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6911>
2024-05-23 21:36:37 +00:00
Jan Schmidt
64133b40a7 rtpmp4gdepay: Set duration on outgoing buffers
If we have constant duration buffers, set the duration on
outgoing buffers, like rtpmp4adepay does. This fixes
problems with (for example) muxers like mp4mux not writing
the duration of the final sample into the index.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6878>
2024-05-20 15:24:32 +00:00
Guillaume Desmottes
210487b50a wavparse: reset when receiving STREAM_START
We need to reset the internal state to be able to parse a new stream.
When doing so keep seek event and do not destroy the adapter.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6840>
2024-05-16 11:35:02 +00:00
Sebastian Dröge
a4514c5458 level: Don't post a message on EOS without a valid audio info
If EOS is received before caps, e.g. because of an error, then rate and
number of channels would be 0 and some divisions by zero would happen.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6819>
2024-05-12 07:06:32 +00:00
Sebastian Dröge
0ef396359c gst: Move GstQueueArray as GstVecDeque to core
And change lengths and indices from guint to gsize for a more correct type.

Also deprecate GstQueueArray and implement it in terms of GstVecDeque.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6779>
2024-05-06 18:25:42 +00:00
Tim-Philipp Müller
eec64e372b rtph264depay: fix FU-B handling
Skip extra 16-bit DON in FU-B header.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/806

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6607>
2024-04-29 12:21:52 +00:00
Tim-Philipp Müller
b1a45b527a rtph264depay: minor refactoring of FU handling code
Make code easier to follow, and prepare for next commit.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6607>
2024-04-29 12:21:52 +00:00
Qian Hu (胡骞)
cd95d02032 qtdemux: fix wrong full_range offset when parsing colr box
use colr_data[18] >> 7 to get full range information, instead
of colr_data[17] >> 7

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6616>
2024-04-12 16:59:33 +08:00
Sebastian Dröge
0596871b98 rtpbin: Don't re-use a variable for a completely different purpose temporarily
During RTP-Info synchronization, clock_base was temporarily switched
from the actual clock-base to the base RTP time and then back some lines
later.

Instead directly work with the base RTP time. The comment about using a
signed variable for convenience doesn't make any sense because all
calculations done with the value are unsigned.

Similarly, rtp_clock_base was overridden with the rtp_delta when
calculating it, which was fine because it is not used anymore
afterwards. Instead, introduce a new variable `rtp_delta` to make this
calculation clearer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6536>
2024-04-08 10:29:54 +00:00
Sebastian Dröge
11ce209ea0 rtpbin: Convert clock-base to extended RTP timestamp correctly
It's not in the same period as the current RTP base time but always in
the very first period. This avoids using it again at a much later time.

The code in question is only triggered with rtcp-sync=rtp-info.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6536>
2024-04-08 10:29:54 +00:00
Sebastian Dröge
0c34c85f7a rtpjitterbuffer: Use an extended RTP timestamp for the clock-base
It is compared to other extended RTP timestamps all over rtpjitterbuffer
and since 4df3da3bab the initial extended RTP timestamp is not equal
anymore to the plain RTP time.

Continue passing a non-extended RTP timestamp via the `sync` signal for
backwards compatibility. It will always be a timestamp inside the first
extended timestamp period anyway.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6536>
2024-04-08 10:29:54 +00:00
Sebastian Dröge
4a4eb56fc2 rtspsrc: Optionally timestamp RTP packets with their receive times in TCP/HTTP mode
Until https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6509
this was accidentally done inside rtpjitterbuffer for many years, and
doing so potentially solves problems on some streams while introducing
problems on others.

Make this configurable on rtspsrc and default to not set timestamps.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6529>
2024-04-08 08:34:38 +00:00
Jan Schmidt
832a517965 rtpjitterbuffer: Don't use estimated_dts to do default skew adjustment
When the buffer DTS is estimated based on arrival time at the
jitterbuffer (rather than provided on the incoming buffer itself),
it shouldn't be used for skew adjustment. The typical case is
packets being deinterleaved from a tunnelled TCP/HTTP RTSP stream,
and the arrival times at the jitter buffer are not well enough
correlated to usefully do skew adjustments.

This restores the original intended behaviour for the 'estimated dts'
path, that was broken years ago during other jitterbuffer refactoring.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6509>
2024-04-07 12:24:58 +00:00
Sebastian Dröge
16f69acf30 wavpackparse: Use an unsigned integer for the block size calculations
It's never negative.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6498>
2024-04-04 15:10:02 +00:00
Sebastian Dröge
eefb7c1638 wavpackparse: Fix potential integer overflow on ID_ODD_SIZE blocks
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6498>
2024-04-04 15:10:02 +00:00
Sebastian Dröge
6402978a48 wavpackparse: Explicitly handle ID_WVX_NEW_BITSTREAM
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6498>
2024-04-04 15:10:02 +00:00
Jan Schmidt
351936aeac rtpmp4adepay: Set duration on outgoing buffers
If we can calculate timestamps for buffers, then set the duration
on outgoing buffers based on the number of samples depayloaded.

This can fix the muxing to mp4, where otherwise the last packet
in a muxed file will have 0 duration in the mp4 file.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6447>
2024-03-27 10:53:38 +00:00
Sebastian Dröge
e0dfb3d974 rtphdrext-ntp: Fix typo of the RFC number in the element metadata
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3417

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6439>
2024-03-26 14:37:47 +02:00
Alexander Slobodeniuk
650534c940 rtspsrc: remove 'deprecated' flag from the 'push-backchannel-sample' signal
It seems that it was added by accident when copying from push-backchannel-buffer

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6355>
2024-03-13 19:32:46 +00:00
Piotr Brzeziński
9c084faa75 qtdemux: Fix wrapping temporary memory in buffers
That memory can disappear at any moment, doesn't cost much to just copy those few bytes.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6320>
2024-03-11 18:18:01 +00:00
Nirbheek Chauhan
3bed35c342 rtspsrc: Don't invoke close when stopping if we've started cleanup
When we're doing a state change from PLAYING to NULL, first we invoke
gst_rtspsrc_loop_send_cmd_and_wait (..., CMD_CLOSE, ...) during
PAUSED_TO_READY which will schedule a TEARDOWN to happen async on the
task thread.

The task thread will call gst_rtspsrc_close(), which will send the
TEARDOWN and once it's complete, it will call gst_rtspsrc_cleanup()
without taking any locks, which frees src->streams.

At the same time however, the state change in the app thread will
progress further and in READY_TO_NULL it will call gst_rtspsrc_stop()
which calls gst_rtspsrc_close() a second time, which accesses
src->streams (without a lock again), which leads to simultaneous
access of src->streams, and a segfault.

So the state change and the cleanup are racing, but they almost always
complete sequentially. Either the cleanup sets src->streams to NULL or
_stop() completes first. Very rarely, _stop() can start while
src->streams is being freed in a for loop. That causes the segfault.

This is unlocked access is unfixable with more locking, it just leads
to deadlocks. This pattern has been observed in rtspsrc a lot: state
changes and cleanup in the element are unfixably racy, and that
foundational issue is being addressed separately via a rewrite.

The bandage fix here is to prevent gst_rtspsrc_stop() from accessing
src->streams after it has already been freed by setting src->state to
INVALID.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6302>
2024-03-11 09:15:50 +00:00
François Laignel
7d5bb1ea7a webrtc: add all SSRC attributes getting CAPS for a PT
The transport stream only returned the CAPS for the first matching PT entry
from the `ptmap`. Other SSRC with the same PT where not included. For a stream
which bundled multiple audio streams for instance, only the first SSRC was
knowed to the SSRC demux and downstream elements.

This commit adds all the `ssrc-` attributes from the matching PT entries.

The RTP jitter buffer can now find the CNAME corresponding its SSRC even if it
was not the first to be registered for a particular PT.

The RTP PT demux removes `ssrc-*` attributes cooresponding to other SSRCs
before pushing SSRC specific CAPS to downstream elements.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6119>
2024-03-08 10:28:15 +00:00
Mathieu Duponchelle
519546aea3 rtpgstpay: flush on EOS
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5173>
2024-03-07 14:02:33 +00:00
Sebastian Dröge
b88d69b722 rtpgstpay: Delay pushing of event packets until the next buffer
And also re-timestamp them with the current buffer's PTS.

Not doing so keeps the timestamps of event packets as
GST_CLOCK_TIME_NONE or the timestamp of the previous buffer, both of
which are bogus.

Making sure that (especially) the first packet has a valid timestamp
allows putting e.g. the NTP timestamp RTP header extension on it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5173>
2024-03-07 14:02:33 +00:00
Elizabeth Figura
e2167867d5 qtdemux: Do not set channel-mask to zero
Leave it uninitialized, so that the downstream decoder will initialize it appropriately. Setting it to zero is wrong.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6225>
2024-03-07 12:52:30 +02:00
Jan Schmidt
f53dbb28b2 rtspsrc: Parse Speed/Scale before Range in responses
Parse the speed and scale in the server's response
*before* the range, so that the range start/stop
are swapped (or not swapped) correctly based
on the server's actual chosen values. Otherwise,
the old rate from the segment is used - what the
last seek asked for, but not necessarily what
the server chooses.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6248>
2024-03-06 17:50:53 +00:00
Jan Schmidt
57013e1a7c rtspsrc: Handle queries and events with no manager
When doing direct output with no session manager, we still
want to respond to queries and events from downstream, so
install the handlers

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6248>
2024-03-06 17:50:53 +00:00
Jan Schmidt
4d2f000125 rtspsrc: return NO_PREROLL on PLAYING->PAUSED too
When transitioning back to PAUSED and rtspsrc is live, return
NO_PREROLL so the pipeline knows to skip preroll here too.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6248>
2024-03-06 17:50:53 +00:00
Tim-Philipp Müller
4db25f1500 rtspsrc: Consider 503 Service Not Available when handling broken control urls
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6213>
2024-03-05 17:45:18 +00:00
Nirbheek Chauhan
cf2238a522 rtspsrc: Increase rank to PRIMARY for autoplug purposes
This affects autoplug by gst_element_make_from_uri() in, for example,
uridecodebin. The element should've already been PRIMARY rank, but it
was NONE because gst_element_make_from_uri() doesn't ignore NONE rank
elements when searching for element factories, unlike decodebin.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/502

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6226>
2024-02-27 11:36:01 +00:00
Sebastian Dröge
69e4564c87 rtphdrext-clientaudiolevel: Fix typo in documentation
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6175>
2024-02-21 17:25:43 +00:00
Tim-Philipp Müller
0a6948ee20 rtppassthroughpay: fix critical in gst-inspect
gst_segment_to_running_time() will fail noisily
if the segment has not been initialised yet.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6151>
2024-02-21 11:25:10 +00:00
Jan Schmidt
f7e494f348 rtspsrc: Reset combined flows after a seek before restarting
After a flushing seek, rtspsrc doesn't reset the last_ret value for
streams, so might immediately shut down again when it resumes pushing
buffers to pads due to a cached `GST_FLOW_FLUSHING` result

Prevent a stored flushing value from immediately stopping
playback again by resetting pad flows before (re)starting
playback.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6137>
2024-02-21 01:50:13 +00:00
Jochen Henneberg
6608b89977 rtpxqtdepay: Enabled header extension aggregation
Because this depayloader may build several output buffers within one
process run we push them all into a GstBufferList and push them out at
once to make sure that each buffer gets notified about each header
extension.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5378>
2024-02-19 11:23:17 +00:00
Jochen Henneberg
5d1d0cf9a5 rtpmp4gdepay: Enabled header extension aggregation
Because this depayloader may build several output buffers within one
process run we push them all into a GstBufferList and push them out at
once to make sure that each buffer gets notified about each header
extension.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5378>
2024-02-19 11:23:17 +00:00
Jochen Henneberg
75849c63c8 rtpsbcdepay: Enabled header extension aggregation
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5378>
2024-02-19 11:23:17 +00:00