Commit graph

124 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
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
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
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
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
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
Mathieu Duponchelle
91317aacaf webrtcbin, rtpbin: check before setting properties on jitterbuffer
In rtpbin we already systematically check for all property names
except latency, correct that.

In webrtcbin we need to check before trying to use the do-retransmission
property.

This is useful for the case where an element like identity gets passed
to rtpbin's request-jitterbuffer property, when the application wants
to use webrtcbin in an SFU situation, with no reordering and no added
latency

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6112>
2024-02-14 08:52:50 +00:00
Sebastian Dröge
c726add352 rtpfunnel: Handle NTP-64 RTP header extension in caps similar to TWCC
This is another header extension that is handled by rtpsession and needs
to be preserved in the caps that are created by rtpfunnel.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6109>
2024-02-14 08:05:33 +00:00
Sebastian Dröge
17e7af7181 rtpfunnel: Also write TWCC RTP header extension into buffer list buffers
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6110>
2024-02-14 01:56:20 +00:00
Hou Qi
2539bb0b1d rtpjitterbuffer: Fix build warning in rtp_jitter_buffer_append_query()
This is to fix build warnings when using [-Wmaybe-uninitialized]
../gst/rtpmanager/rtpjitterbuffer.c:1237:10: warning: 'head' may be used uninitialized [-Wmaybe-uninitialized]
 1237 |   return head;

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5907>
2024-01-13 15:00:19 +00:00
Sebastian Dröge
6fa41f78bb rtpsession: Remove some unused fields
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5899>
2024-01-08 12:57:04 +02:00
Sanchayan Maity
00bbac6541 rtphdrext-clientaudiolevel: Fix level value being written by the extension
When level value is greater than 127, it was being clamped but this clamped
value was not the one being actually used. For level values greater than 127
this resulted in an incorrect value being used. As an example, a level value
of 187, after and'ed with 0x7F, it would result in 0x3B being reported as the
level value.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5893>
2024-01-07 16:00:18 +05:30
Sebastian Dröge
c292da7044 rtpsession: Only warn once if configured latency needs to be known but isn't yet
Otherwise we would warn about this once for every single packet until
the LATENCY event is received.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5854>
2023-12-27 11:00:44 +00:00
Sebastian Dröge
db77deef00 rtpjitterbuffer: Add new "rfc7273-reference-timestamp-meta-only" property
If this property is enabled then the jitterbuffer will do the normal PTS
calculations according to the configured mode instead of making use of
the RFC7273 media clock.

The timestamp calculated from the RFC7273 media clock will only be
stored in the reference timestamp meta, if addition of that meta is enabled.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5512>
2023-11-16 15:23:29 +00:00
Sebastian Dröge
eae3ef7461 rtpjitterbuffer: Add new rfc7273-use-system-clock property
When this property is used, it is assumed that the system clock is
synced close enough to the media clock used by an RFC7273 stream.

As long as both clocks are at most a few seconds from each other this
will give the correct results and avoids having to create an actual
network clock that has to sync first.

If the system clock is actually synchronized to the media clock then
everything will behave exactly the same, otherwise the reference
timestamp meta will be correct but the buffer timestamps will be off by
the difference between the two clocks.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5512>
2023-11-16 15:23:29 +00:00
Sebastian Dröge
2956ba48fc rtpjitterbuffer: Improve handling of media clocks
Do more checks for clock equality than just checking pointers. The same
NTP/PTP clock might be used as pipeline clock but a new instance, so
instead also check what clock they are synced to.

Also handling setting / resetting of the media clock and pipeline clock
correctly by resetting the media clock's state accordingly.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5512>
2023-11-16 15:23:29 +00:00
Sebastian Dröge
2a2ef23829 rtpsource: Don't store invalid running times and calculate with it
If we end up with GST_CLOCK_TIME_NONE as running time for an RTP packet
then this can't be used for bitrate estimation, and also not for
constructing the next RTCP SR. Both would end up with completely wrong
values, and an RTCP SR with wrong values can easily break
synchronization in receivers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5329>
2023-09-23 07:39:00 +00:00
Sebastian Dröge
fcd591c1af rtpjitterbuffer: Avoid integer overflow in max saveable packets calculation with negative offset
The timestamp offset can be negative, and it can be a bigger negative
number than the latency introduced by the rtpjitterbuffer so the overall
timeout offset can be negative.

Using the negative offset for calculating how many packets can still
arrive in time when encountering a lost packet in an equidistant stream
would then overflow and instead of considering fewer packets lost a lot
more packets are considered lost.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5296>
2023-09-12 08:38:53 +00:00
Charlie Blevins
05cffc19dd rtpjitterbuffer: Allow earlier reference-timestamp-meta
Allow reference-timestamp-meta to be added earlier if an RTCP sender
report is sent before the first RTP packet.

Fixes #2843

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5084>
2023-08-03 17:26:42 +00:00
Mathieu Duponchelle
7445b73e76 rtpsession: expose timeout-inactive-sources property
In some situations it is not desirable to time out when no data is
received any longer, users can opt in to this behavior via a new
property.

The property is also exposed on rtpbin and sdpdemux

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4880>
2023-06-28 18:45:25 +00:00
Sebastian Dröge
f9a3b3eacf rtpjitterbuffer: Fix uninitialized variable compiler warning
It could indeed be used uninitialized, but only if one of the
g_return_val_if_fail() caused an early return.

../subprojects/gst-plugins-good/gst/rtpmanager/rtpjitterbuffer.c: In function ‘rtp_jitter_buffer_append_query’:
../subprojects/gst-plugins-good/gst/rtpmanager/rtpjitterbuffer.c🔢10: warning: ‘head’ may be used uninitialized
      [-Wmaybe-uninitialized]
 1234 |   return head;
      |          ^~~~
../subprojects/gst-plugins-good/gst/rtpmanager/rtpjitterbuffer.c:1232:12: note: ‘head’ was declared here
 1232 |   gboolean head;
      |            ^~~~

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4616>
2023-05-14 14:26:05 +00:00
François Laignel
6675ed9aae rtpmanager/rtsession: data race leading to critical warnings
This is a fix for a data race leading to:

> GLib-CRITICAL: g_hash_table_foreach:
>   assertion 'version == hash_table->version' failed

Identified sequence:

* `rtp_session_on_timeout` acquires the lock on `session` and proceeds with its
  processing.
* `rtp_session_process_rtcp` is called (debug log : received RTCP packet) and
  attempts to acquire the lock on `session`, which is still held by
  `rtp_session_on_timeout`.
* as part of an hash table iterator, `rtp_session_on_timeout` transitively
  invokes `source_caps` which releases the lock on `session` so as to call
  `session->callbacks.caps`.
* Since `rtp_session_process_rtcp` was waiting for the lock to be released, it
  succeeds in acquiring it and proceeds with `rtp_session_process_rr` which
  transitively calls `g_hash_table_insert` via `add_source`.
* After `source_caps` re-acquires the lock and gives the control flow back to
  `rtp_session_on_timeout`, the hash table iterator is changed, resulting in the
  assertion failure.

This commits copies `sess->ssrcs[sess->mask_idx]` and iterates on the copy so
the iterator is not affected by a concurrent change due to the lock being
released in the `source_caps` callback.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4555>
2023-05-09 16:05:29 +00:00
François Laignel
5ef2ce69ff rtpmanager/rtsession: race conditions leading to critical warnings
While testing the [implementation for insertable streams] in `webrtcsink` &
`webrtcsrc`, I encountered critical warnings, which turned out to result from
two race conditions in `rtpsession`. Both race conditions produce:

> GLib-CRITICAL: g_hash_table_foreach:
>   assertion 'version == hash_table->version' failed

This commit fixes one of the race conditions observed.

In its simplest form, the test consists in 2 pipelines and a Signalling server:

* pipelines_sink: audiotestsrc ! webrtcsink
* pipelines_src: webrtcsrc ! appsrc

1. Set `pipelines_sink` to `Playing`.
2. The Signalling server delivers the `producer_id`.
3. Initialize `pipelines_src` to establish a session with `producer_id`.
4. Set `pipelines_src` to `Playing`.
5. Wait for a buffer to be received by the `appsrc`.
6. Set `pipelines_src` to `Null`.
7. Set `pipelines_sink` to `Null`.

The race condition happens in the following sequence:

* `webrtcsink` runs a task to periodically retrieve statistics from `webrtcbin`.
  This transitively ends up executing `rtp_session_create_stats`.
* `pipelines_sink` is set to `Null`.
* In `Paused` to `Ready`, `gst_rtp_session_change_state()` calls
  `rtp_session_reset()`.
* The assertion failure occurs when `rtp_session_reset` is called while
  `rtp_session_create_stats` is executing.

This is because `rtp_session_create_stats` acquires the lock on `session` prior
to calling `g_hash_table_foreach`, but `rtp_session_reset` doesn't acquire the
lock before calling `g_hash_table_remove_all`.

Acquiring the lock in `rtp_session_reset` fixes the issue.

[implementing insertable streams support]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1176

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4528>
2023-05-02 21:56:39 +00:00
Mathieu Duponchelle
6a27fe8955 docs: mark GstRTPMux as plugin API
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4408>
2023-04-13 21:46:59 +00:00
Edward Hervey
7e619f7e83 twcc: Better handle duplicate packets
The previous code would only check if two packets in a row were duplicates. If
not (i.e. a packet is a duplicate of a packet received slightly before) the code
would generate completely bogus FCI because it assumes there were no duplicates
present in the array.

In order to be efficient, just store all received packets and remove the
duplicates just before the FCI is generated once the array of observations have
been sorted by seqnum.

Fixes TWCC usage with moderate to high packet duplication.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4328>
2023-04-10 09:37:51 +00:00
Scott Kanowitz
2e4fd325e7 rtpsession: fix a race condition during the EOS event in gstrtpsession.c
This patch prevents a possible race condition from taking place between the EOS event handling and rtcp send
function/thread.

The condition starts by getting the GST_EVENT_EOS event on the send_rtp_sink pad, which causes two core things
to happen -- the event gets pushed down to the send_rtp_src pad and all sessions get marked "bye" prior to
completion of the event handler. In another thread the rtp_session_on_timeout function gets called after an
expiration of gst_clock_id_wait in the rtcp_thread function. This results in a call to the
ess->callbacks.send_rtcp(), which is configured as a function pointer to gst_rtp_session_send_rtcp via the
RTPSessionCallbacks structure passed to rtp_session_set_callbacks in the gst_rtp_session_init function.

In the race condition, the call to gst_rtp_session_send_rtcp can have the all_sources_bye boolean set to true
while GST_PAD_IS_EOS(rtpsession->send_rtp_sink) evaluates to false. This is the result of gst_rtp_session_send_rtcp
running before the send_rtp_sink's GST_EVENT_EOS handler completes. The exact point at which this condition occurs
is if there's a context switch to the rtcp_thread right after the call to rtp_session_mark_all_bye in the
GET_EVENT_EOS handler, but before the handler returns.

Normally, this would not be an issue because the rtcp_thread continues to run and indirectly call
gst_rtp_session_send_rtcp. However, the call to rtp_source_reset sets the sent_bye boolean to false, which ends up
causing rtp_session_are_all_sources_bye to return false. This gets passed to gst_rtp_session_send_rtcp and the EOS
event never gets sent.

The race condition results in the EOS event never getting passed to the rtcp_src pad, which prevents the bin and
pipeline from ever completing with EOS.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3798>
2023-02-28 17:01:08 +00:00
Guillaume Desmottes
3d1390d31a rtpptdemux: set different stream-id on each src pad
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3855>
2023-02-01 09:17:33 +00:00
Guillaume Desmottes
cc2b8f6ae8 rtpssrcdemux: set different stream-id on each src pad
All the RTP src pads were sharing the same stream-id while each actually
carry a different stream.

This was causing problem for example when funneling the streams together
and then trying to split them using 'streamiddemux'.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3855>
2023-02-01 09:17:33 +00:00
Tim-Philipp Müller
8222b97331 rtpmanager: drop use of GSlice
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3695>
2023-01-24 15:25:06 +00:00
Seungha Yang
9b305df1cc rtptimerqueue: Fix memory leak
Should chain up to parent's finalize

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3645>
2022-12-27 19:31:16 +00:00
Matt Crane
ca7f66f9b5 rtpsession: Support disabling late adjustment of ntp-64 header ext
Currently in rtp_session_send_rtp(), the existing ntp-64 RTP header
extension timestamp is updated with the actual NTP time before sending
the packet. However, there are some circumstances where we would like
to preserve the original timestamp obtained from reference timestamp
buffer metadata.

This commit provides the ability to configure whether or not to update
the ntp-64 header extension timestamp with the actual NTP time via the
update-ntp64-header-ext boolean property. The property is also exposed
via rtpbin. Default property value of TRUE will preserve existing
behavior (update ntp-64 header ext with actual NTP time).

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1580

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3451>
2022-11-24 08:23:03 +00:00
Johan Sternerup
9794c9bfd0 Use the correct SSRC(s) when routing a RTCB FB FIR
Previously we tried to route an incoming RTCP FB FIR to the correct ssrc
using the "media source" component of the RTCP FB message. However,
according to RFC5104 (section 4.3.1.2) the "media source" SHALL be set
to 0. Instead the ssrc(s) in use are propagated via the FCI data. Now
a specific GstForceKeyUnit event is sent for every ssrc.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3292>
2022-11-23 11:31:23 +00:00
Jan Schmidt
cb225b3682 rtpsource: Track the seqnum for senders
RTP source statistics are tracked for local senders by
treating them as a receiver of their own outbound packets.

Accordingly, track the highest packet seqnum so that the
packets-lost calculation generates a sensible number instead
of always reporting -$number_of_packets_sent

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3454>
2022-11-23 10:26:29 +00:00
Sebastian Dröge
3d79402344 rtpjitterbuffer: Reschedule timers when updating their offset
As EXPECTED timers are skipped the order of the timers relative to each
other can change if there are EXPECTED timers and rescheduling needs to
happen.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1422

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3416>
2022-11-16 08:26:41 +00:00
Edward Hervey
30886fa9ea rtpjitterbuffer: Unlock timer waits on flushing
If there is a pending EOS wait for example, we would never unblock on flushing

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3401>
2022-11-15 18:30:43 +00:00
Sebastian Dröge
bd5a4d321b rtpsource: Don't do probation for RTX sources
Disable probation for RTX sources as packets will arrive very
irregularly and waiting for a second packet usually exceeds the deadline
of the retransmission.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/181

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3112>
2022-10-10 14:56:18 +00:00