Commit graph

1446 commits

Author SHA1 Message Date
Havard Graff
f997859913 rtpsession: add locking for clear-pt-map
...or it will segfault from time to time...
2019-11-29 14:23:49 +01:00
Niels De Graef
7cf4ab6229 Don't pass default GLib marshallers for signals
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.
2019-11-17 15:32:30 +00:00
Nicolas Dufresne
db187eec19 rtpjitterbuffer: Check the exit condition after executing timers
The do_expected_timeout() function may release the JBUF_LOCK, so we need
to check if nothing wanted the timer thread to exit after this call.
The side effect was that we may endup going back into waiting for a timer
which will cause arbitrary delay on tear down (or deadlock when test
clock is used).

Fixes #653
2019-11-14 17:52:16 -05:00
Nicolas Dufresne
fd6cd6f545 rtpjitterbuffer: Check exit condition immediately after JBUF_WAIT
JBUF_WAIT_QUEUE drops the JBUF_LOCK, which means the stop condition
for the chain function may have changed (change_state to NULL). Check
this immediately after the wait so that we don't delay shutting down.
2019-11-14 17:51:31 -05:00
Havard Graff
87457a862d rtpjitterbuffer: make sure not to drop packets based on skew
One of the jitterbuffers functions is to try and make sense of weird
network behavior.

It is quite unhelpful for the jitterbuffer to start dropping packets
itself when what you are trying to achieve is better network resilience.

In the case of a skew, this could often mean the sender has restarted
in some fashion, and then dropping the very first buffer of this "new"
stream could often mean missing valuable information, like in the case
of video and I-frames.

This patch simply reverts back to the old behavior, prior to 8d955fc32b
and includes the simplest test I could write to demonstrate the behavior,
where a single packet arrives "perfectly", then a 50ms gap happens,
and then two more packets arrive in perfect order after that.

# Conflicts:
#	tests/check/elements/rtpjitterbuffer.c
2019-11-02 23:05:32 +00:00
Edward Hervey
8e1c224fbc good: Avoid usage of deprecated API
GTimeval and related functions are now deprecated in glib.
Replacement APIs have been present since 2.26
2019-10-16 07:46:58 +00:00
Tim-Philipp Müller
c9a47c0c8d Remove autotools build system 2019-10-14 11:04:18 +01:00
Aaron Boxer
46989dca96 documentation: fix a number of typos 2019-10-05 22:38:11 +00:00
Simon Arnling Bååth
8173596ed2 gstrtpjitterbuffer: Custom messages when dropping packets
This commit adds custom element messages for when gstrtpjitterbuffer
drops an incoming rtp packets due to for example arriving too late.
Applications can listen to these messages on the bus which enables
actions to be taken when packets are dropped due to for example high
network jitter.

Two properties has been added, one to enable posting drop messages and
one to set a minimum time between each message to enable throttling the
posting of messages as high drop rates.
2019-10-04 20:31:56 +00:00
Olivier Crête
a24596423a rtpjitterbuffer: Cancel timers instead of just unlocking loop thread
When the queue is full (and adding more packets would risk a seqnum
roll-over), the best approach is to just start pushing out packets
from the other side.  Just pushing out the packets results in the
timers being left hanging with old seqnums, so it's safer to just
execute them immediately in this case. It does limit the timer space
to the time it takes to receiver about 32k packets, but without
extended sequence number, this is the best RTP can do.

This also results in the test no longer needed to have timeouts or
timers as pushing packets in drives everything.

Fixes #619
2019-09-28 07:47:54 -04:00
Nicolas Dufresne
4a9f42430a rtpjitterbuffer: Optimize offset update
As we are applying the same offset over all timers, there timer
ordering won't change, so we can safely skip time-reordering.
2019-09-27 17:34:04 -04:00
Nicolas Dufresne
af1c586c7b rtptimerqueue: Optimize reschedule optations
This basically add ability to choose between inserting from head, tail
or in-place in order to try and minimize the distance to walk through in
the timer queue. This removes an overhead we had seen on high drop rate.
2019-09-27 17:34:04 -04:00
Nicolas Dufresne
1897c1fbe6 rtpjitterbuffer: Fix a typo in comment 2019-09-27 17:34:04 -04:00
Nicolas Dufresne
9ebcadb349 rtpjitterbuffer: Don't use stats timer on the timers queue
The timer passed to update_timers may be from the stats timer. At the
moment, we could endup rescheduling (reusing) that timer onto the normal
timer queue, unschedul it as if it was from the normal timer queue or
duplicate it into the stats timer queue again. This was protected before
as the with the fact the stats timer didn't have a valid idx.
2019-09-27 17:34:04 -04:00
Nicolas Dufresne
81bffb5e5c rtpjitterbuffer: Update timers on ts-offset changes
As the offset is already applied now, we need to update and reschedule
all timers each time the offset is changed. I'm not sure who expect this
to be retro-actively applied, but there was a unit test for it.
2019-09-27 17:34:04 -04:00
Nicolas Dufresne
d4c6c335c5 rtpjitterbuffer: No need to wake the timer thread on head changes
If the jitterbuffer head change, there is no need to systematically
wakeup the timer thread. The timer thread will be waken up on if
an earlier timeout has been pushed. This prevent some more spurious
wakeup when the system is loaded. As a side effect, cranking the clock
may set the clock at an earlier position.
2019-09-27 17:34:04 -04:00
Nicolas Dufresne
36771b75e9 rtpjittterbuffer: Port timers array to RtpTimerQueue
In this patch we now make use of the new RtpTimerQueue instead of the
old GArray. This required a lot of changes all over the place, some of
the important changes are that `timer->timeout` is no longer a PTS but
the actual timeout. This was required to get the RtpTimerQueue sorting
right. The applied offset is saved as `timer->offset`, this allow
retreiving back the PTS when needed.

The clockid updates only happens once per incoming packet. If the
currently schedule timer is before the earliest timer in the queue, we
no longer wakeup the thread. This way, if other timers get setup in the
meantime, this will reduce the number of wakup.

The timer loop code has been mostly rewritten, though the behaviour of
running the lost timers first has been kept (even though there is no
test to show what would be the side effect of doing this differently).

Fixes #608
2019-09-27 17:34:04 -04:00
Nicolas Dufresne
d4b2231de2 rtpjittterbuffer: Port from TimerQueue to RtpTimerQueue 2019-09-27 17:34:04 -04:00
Nicolas Dufresne
f5e3280dbe rtpjitterbuffer: Port use the new RtpTimer structure
First iteration toward porting to the new timer queue.
2019-09-27 17:34:04 -04:00
Nicolas Dufresne
37742cd36d rtptimerqueue: Consolidate a data structure for timers
Implement a single timer queue for all timers. The goal is to always use
ordered queues for storing timers. This way, extracting timers for
execution becomes O(1). This also allow separating the clock wait
scheduling from the timer itself and ensure that we only wake up the
timer thread when strictly needed.

The knew data structure is still O(n) on insertions and reschedule,
but we now use proximity optimization so that normal cases should be
really fast. The GList structure is also embeded intot he RtpTimer
structure to reduce the number of allocations.
2019-09-27 17:34:04 -04:00
Nicolas Dufresne
c917f11ae8 rtpjitterbuffer: Move item structure outside of the element
This moves the RtpJitterBufferStructure type, alloc, free into
rtpjitterbuffer.c/h implementation. jitterbuffer.c strictly rely on
the fact this structure is compatible with GList, and so it make more
sense to keep encapsulate it. Also, anything that could possibly
reduce the amount of code in the element is a win.

In order to support that move, a function pointer to free the data
was added. This also allow making the free function option when
flushing the jitterbuffer.
2019-09-27 13:02:16 -04:00
Nicolas Dufresne
9b706b6220 rtpjitterbuffer: Constify timer pointers where possible
This helps understanding which function modify the Timerdata
and which one does not. This is not always obvious from thelper
name considering recalculate_timer() does not.
2019-09-27 13:02:16 -04:00
Mathieu Duponchelle
b5e414cdc2 rtpbin: add request-jitterbuffer signal
This can be used to pass the threadsharing jitterbuffer from
gst-plugins-rs for example.
2019-09-24 15:33:21 +00:00
Antonio Ospite
8dd03042cc rtpsession: add support for buffer lists on the recv path
The send path in rtpsession processes the buffer list along the way,
sharing info and stats between packets in the same list, because it
assumes that all packets in a buffer list are from the same frame.

However, in the receiving path packets can arrive in all sorts of
arrangements:

  - different sources,
  - different frames (different timestamps),
  - different types (multiplexed RTP and RTCP, invalid RTP packets).

so a more general approach should be used to correctly support buffer
lists in the receive path.

It turns out that it's simpler and more robust to process buffers
individually inside the rtpsession element even if they come in a buffer
list, and then reassemble a new buffer list when pushing the buffers
downstream.

This avoids complicating the existing code to make all functions
buffer-list-aware with the risk of introducing regressions,

To support buffer lists in the receive path and reduce the "push
overhead" in the pipeline, a new private field named processed_list is
added to GstRtpSessionPrivate, it is set in the chain_list handler and
used in the process_rtp callback; this is to achieve the following:

  - iterate over the incoming buffer list;
  - process the packets one by one;
  - add the valid ones to a new buffer list;
  - push the new buffer list downstream.

The processed_list field is reset before pushing a buffer list to be on
the safe side in case a single buffer was to be pushed by upstream
at some later point.

NOTE:

The proposed modifications do not change the behavior of the send path.

The process_rtp callback is called in rtpsource.c by the push_rtp
callback (via source_push_rtp) only when the source is not internal.

So even though push_rtp is also called in the send path, it won't end up
using process_rtp in this case because the source would be internal in
the send path.

The reasoning from above may suggest a future refactoring: push_rtp
might be split to better differentiate the send and receive path.
2019-08-07 15:32:30 -04:00
Mathieu Duponchelle
e18d5d6ec6 rtpfunnel: forward correct segment when switching pad
Forwarding a single segment event from the pad that first gets
chained is incorrect: when that first event was sent by an element
such as x264enc, with its offset start, we end pushing out of segment
buffers for the other pad(s).

Instead, everytime the active pad changes, forward the appropriate
segment event.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/1028
2019-08-06 14:02:50 +00:00
Antonio Ospite
ae48646d8e rtpsource: fix receiver source stats to consider previously queued packets
When it is not clear yet if a packet relative to a source should be
pushed, the packet is put into a queue, this happens in two cases:

  - the source is still in probation;
  - there is a large jump in seqnum, and it is not clear what
    the cause is, future packets will help making a guess.

In either case stats about received packets are not updated at all; and
even if they were, when init_seq() is called it resets all receiver
stats, effectively loosing any possible stat about previously received
packets.

Fix this by taking into account the queued packets and update the stats
when calling init_seq().
2019-08-02 17:22:51 +02:00
Antonio Ospite
cf0ffd8693 rtpsource: clarify meaning of the octets-sent and octets-received stats
The octets-send and octets-received stats count the payload bytes
excluding RTP and lower level headers, clarify that in the
documentation.
2019-08-02 17:22:51 +02:00
Antonio Ospite
821994240e rtpsource: expose field bytes_received in RTPSourceStats
Since commit c971d1a9a (rtpsource: refactor bitrate estimation,
2010-03-02) bytes_received filed in RTPSourceStats is set but then never
used again, expose it so that it can be used  by user code to verify how
many bytes have been received.
2019-08-02 17:22:51 +02:00
Antonio Ospite
9d800cad43 rtpmanager: consider UDP and IP headers in bandwidth calculation
According to RFC3550 lower-level headers should be considered for
bandwidth calculation.

See https://tools.ietf.org/html/rfc3550#section-6.2 paragraph 4:

  Bandwidth calculations for control and data traffic include
  lower-layer transport and network protocols (e.g., UDP and IP) since
  that is what the resource reservation system would need to know.

Fix the source data to accommodate that.

Assume UDPv4 over IP for now, this is a simplification but it's good
enough for now.

While at it define a constant and use that instead of a magic number.

NOTE: this change basically reverts the logic of commit 529f443a6
(rtpsource: use payload size to estimate bitrate, 2010-03-02)
2019-08-02 17:22:51 +02:00
Olivier Crête
9d9d543d5c rtpsession: Also send conflict event when sending packet
If the conflict is detected when sending a packet, then also send an
upstream event to tell the source to reconfigure itself.

Also ignore the collision if we see more than one collision from the same
remote source to avoid problems on loops.
2019-07-06 14:23:20 +00:00
Olivier Crête
37d22186ff rtpjitterbuffer: Unlock output if the queue is full 2019-07-03 18:03:42 +00:00
Thomas Bluemel
080eba64de rtpjitterbuffer: Ignore unsolicited rtx packets.
If an rtx packet arrives that hasn't been requested (it might
have been requested from prior to a reset), ignore it so that
it doesn't inadvertently trigger a clock skew.
2019-07-03 06:23:07 -06:00
Thomas Bluemel
8d955fc32b rtpjitterbuffer: Only calculate skew or reset if no gap.
In the case of reordered packets, calculating skew would cause
pts values to be off. Only calculate skew when packets come
in as expected. Also, late RTX packets should not trigger
clock skew adjustments.

Fixes #612
2019-07-03 06:23:07 -06:00
Olivier Crête
af618cb081 rtpjitterbuffer: max-dropout-time gets cast to int32
So any value over MAXINT32 gets considered as negative and is silently ignored.
2019-07-02 19:59:49 +00:00
Jan Schmidt
53b3f2ddbb rtpjitterbuffer: Clear clock master before unreffing
Make sure to clear any master clock on the media_clock
before unreffing it to release the timer callback that's
updating the clock and keeping it reffed.
2019-06-16 20:36:55 +10:00
Mathieu Duponchelle
ebe2756434 jitterbuffer: unset DTS on output buffers 2019-06-14 16:02:59 +02:00
Mikhail Fludkov
ec5fa49631 rtpjitterbuffer: late packets shouldn't affect PTS of the following packet
If, say, a rtx-packet arrives really late, this can have a dramatic
effect on the jitterbuffer clock-skew logic, having it being reset
and losing track of the current dts-to-pts calculations, directly affecting
the packets that arrive later.

This is demonstrated in the test, where a RTX packet is pushed in really
late, and without this patch the last packet will have its PTS affected
by this, where as a late RTX packet should be redundant information, and
not affect anything.
2019-06-13 11:55:10 +02:00
Mikhail Fludkov
b9c3e354ee rtpjitterbuffer: fix rtx delay calulation when large packet spacing 2019-06-12 11:39:32 +02:00
Stian Selnes
6269ed49ab rtpjitterbuffer: Fix delay for EXPECTED timers added by gaps
This patch corrects the delay set on EXPECTED timers that are added when
processing gaps. Previously the delay could be too small so that
'timout + delay' was much less than 'now', causing the following retries
to be scheduled too early. (They were sent earlier than
rtx-retry-timeout after the previous timeout.)
2019-06-12 11:39:32 +02:00
Havard Graff
8ed7ab178b rtpjitterbuffer: don't try and calculate packet-rate if seqnum are jumping
Turns out that the "big-gap"-logic of the jitterbuffer has been horribly
broken.

For people using lost-events, an RTP-stream with a gap in sequencenumbers,
would produce exactly that many lost-events immediately.
So if your sequence-numbers jumped 20000, you would get 20000 lost-events
in your pipeline...

The test that looks after this logic "test_push_big_gap", basically
incremented the DTS of the buffer equal to the gap that was introduced,
so that in fact this would be more of a "large pause" test, than an
actual gap/discontinuity in the sequencenumbers.

Once the test was modified to not increment DTS (buffer arrival time) with
a similar gap, all sorts of crazy started happening, including adding
thousands of timers, and the logic that should have kicked in, the
"handle_big_gap_buffer"-logic, was not called at all, why?

Because the number max_dropout is calculated using the packet-rate, and
the packet-rate logic would, in this particular test, report that
the new packet rate was over 400000 packets per second!!!

I believe the right fix is to don't try and update the packet-rate if
there is any jumps in the sequence-numbers, and only do these calculations
for nice, sequential streams.
2019-06-12 11:39:31 +02:00
Nicolas Dufresne
f7c712d0b8 rtpssrcdemux: Avoid taking streamlock out-of-band
In this change we now protect the internal srcpads list using the
stream lock and limit usage of the internal stream lock to
preventing data flowing on the other src pad type while creating
and signalling the new pad.

This fixes a deadlock with RTPBin shutdown lock. These two locks would
end up being taken in two different order, which caused a deadlock. More
generally, we should not rely on a streamlock when handling out-of-band
data, so as a side effect, we should not take a stream lock when
iterating internal links.
2019-06-04 09:26:06 -04:00
Vivia Nikolaidou
987230a759 rtpjitterbuffer: Print GstClockTimeDiff as GST_STIME_FORMAT 2019-05-26 17:46:06 +03:00
Mathieu Duponchelle
d704790519 doc: fix element section documentations
Element sections were not rendered anymore after the hotdoc
port, fixing this revealed a few incorrect links.
2019-05-25 16:57:31 +02:00
Nicolas Dufresne
4e0bdca3f0 rtpbin: Improve RTPStorage action signal documentation
This is a tiny clarification as the storage was loosely named "storage".
This change clarify that the storage is specificaly used for received RTP
packets. This is unlike the storage found in rtprtxsend that stores a
backlog of sent RTP packets.
2019-05-25 13:44:00 +02:00
Nicolas Dufresne
947a37f3c8 rtpsession: Always keep at least one NACK on early RTCP
We recently added code to remove outdate NACK to avoid using bandwidth
for packet that have no chance of arriving on time. Though, this had a
side effect, which is that it was to get an early RTCP packet with no
feedback into it. This was pretty useless but also had a side effect,
which is that the RTX RTT value would never be updated. So we we stared
having late RTX request due to high RTT, we'd never manage to recover.

This fixes the regression by making sure we keep at least one NACK in
this situation. This is really light on the bandwidth and allow for
quick recover after the RTT have spiked higher then the jitterbuffer
capacity.
2019-05-17 19:13:22 +00:00
Thibault Saunier
38c5ba90b3 doc: Fix some docstrings 2019-05-13 17:00:00 -04:00
Thibault Saunier
af01988534 doc: Port documentation to hotdoc 2019-05-13 11:34:56 -04:00
Thibault Saunier
232e3682ea Mark some properties as DOC_SHOW_DEFAULT 2019-05-13 10:24:40 -04:00
Thibault Saunier
0a6a62aa76 docs: Port all docstring to gtk-doc markdown 2019-05-13 10:24:40 -04:00
Nicolas Dufresne
a6e7f258ac rtpsource: Add more information to probation warning 2019-05-02 14:44:58 -04:00
Nicolas Dufresne
84c102b6fe rtpsession: Call on-new-ssrc earlier
Right now, we may call on-new-ssrc after we have processed the first
RTP packet. This prevents properly configuring the source as some
property like "probation" are copied internally for use as a
decreasing counter. For this specific property, it prevents the
application from disabling probation on auxiliary sparse stream.

Probation is harmful on sparse streams since the probation algorithm
assume frequent and contiguous RTP packets.
2019-05-02 14:44:58 -04:00
Danny Smith
037d70c01b rtpbin: Free storage when freeing session 2019-04-29 10:57:38 +02:00
Nicolas Dufresne
ec06268ed8 rtpsession: Allow overriding NACK packet creation
This introduce a new signal on RTSession, on-sending-nacks is emited
right before the list of seqnums to be nacked are processed and
transformed into FB Nack. This allow implementing custom nacks
handling through another mechanism with APP feedback.
2019-04-05 18:36:36 -04:00
Mathieu Duponchelle
280d86a841 rtpsession: Add disable-sr-timestamp property
The Onvif Streaming Spec, in section 6.11, mandates that when
Rate-Control is disabled potential RTCP packets shall have
their timestamps set to 0.

<https://www.onvif.org/specs/stream/ONVIF-Streaming-Spec.pdf>
2019-04-05 20:23:08 +02:00
Nicolas Dufresne
6bb53e75fb rtpsession: Send as many nack seqnum as possible
In order to do that, we now split the nacks registration from the actual
FB nack packet construction. We then try and add as many FB Nacks as
possible into the active packets and leave the remaining seqnums in the
RTPSource. In order to avoid sending outdated NACK later on, we save the
seqnum calculated deadline and cleanup the outdated seqnums before the
next RTCP send.

Fixes #583
2019-04-05 14:53:09 +00:00
John Bassett
74a74bfc99 rtpsession: Fix race when sending PLI, FIR and NACK packets
Calling rtp_session_send_rtcp before marking the source as requiring a
pli/fir/nack meant the rtcp_thread could be scheduled and start running
before the source was updated. This meant the request would not be sent
early but instead was transmitted with the next regular RTCP packet.

Add test for nack generation.
2019-04-05 14:53:09 +00:00
Nicolas Dufresne
6b50d142f3 rtpsession: Fix early rtcp time comparision
If the current time is equal to the early rtcp time deadline, there is
no need to schedule a timer. This ensure that immediate feedback is
really immediate and simplify implementing unit tests with the test
clock, which stops perfectly on the timeout time.

This fix has been extracted from Pexip feature patch called
  "rtpsession: Allow instant transmission of RTCP packets"
2019-04-05 14:53:09 +00:00
Antonio Ospite
435f67debf docs: fix typo s/abonormally/abnormally/ 2019-04-03 16:42:26 +02:00
Antonio Ospite
d6939c4031 docs: fix typo s/incomming/incoming/ 2019-04-03 16:38:56 +02:00
Antonio Ospite
114de8cc96 rtpsession: fix comment to refer to buffers instead of groups
One comments in gst_rtp_session_chain_send_rtp_common() is referring to
groups in a buffer list, however this concept of "group" comes from
GStreamer 0.10 and does not exist anymore in GStreamer 1.0, so update the
comment to refer to buffers instead.
2019-04-02 13:03:56 +02:00
Antonio Ospite
e98b0ca8da rtpsource: add comment to explain why probation queue is not always cleared 2019-04-02 13:03:56 +02:00
Antonio Ospite
0fae88b5fd rtpsource: fix stats about received packets
The update_receiver_stats() function is called also when sending packets
in rtp_source_send_rtp(), and sending packets may happen using a buffer
list rather than individual buffers.

So update the stats using the actual number of packets sent.

NOTE: this is fine for the receive path too (rtp_process_send_rtp)
because the receive path does not support buffer lists and
pinfo->packets would always be equal to 1 in this case.
2019-04-02 09:26:03 +02:00
Olivier Crête
0ecc52c2ee rtpbin: Request the FEC decoder even if ignore-pt is set 2019-03-28 16:24:17 -04:00
Olivier Crête
c2dd263562 rtpbin: Factor out the code that exposes the src pad 2019-03-28 16:24:12 -04:00
Nicolas Dufresne
79fd0af152 gstrtpsession: Remove set but not use running-time 2019-03-22 20:01:52 +00:00
Olivier Crête
7ecbd7271d rtpmanager: Register chain functions to debug 2019-03-22 16:44:41 +00:00
Nicolas Dufresne
2ff7519d73 rtpbin: Allow reusing the sender AUX bin
This is needed for the case you don't know in advance all the sessions
you will be using, but would like to place all the related AUX element
in the same GstBin. As per current implementation, each time an sender
AUX bin is requested and returned, RTPBin will walk the src pads and
create sessions for these pads.

In the current implementation, if a src pad already have a sessions, it
returns an error and stops. As a side effect, if an AUX bin is reused in
a following AUX bin request, it can only work if the pads are created on
the last request.

This change simply relax the restriction in order to keep walking, and
just ensure that all newly created pads have a sessions.
2019-03-21 21:10:43 +00:00
George Kiagiadakis
d5ce10240a gstrtpsession: improve stats about rtx requests 2019-03-21 13:40:31 -04:00
George Kiagiadakis
db647ee55b rtprtxsend: Improve looging of not found RTX packet
When an RTX packet is not found, display a message that say if the
packet have not arrived yet or if it was already removed from the RTX
packet queue.
2019-03-21 13:19:52 -04:00
Nicolas Dufresne
0aff8a7d30 rtpsession: Remove unused rtp_session_create_source 2019-03-21 13:19:52 -04:00
Antonio Ospite
30db93e3a4 rtpsource: fix documentation of rtp_source_send_rtp parameters
In commit 28e5f9098 (rtpbin: use PacketInfo for the sender, 2013-09-13)
the rtp_source_send_rtp signature changed but the documentation was not
adjusted to match the new one.

Update the documentation to match the function signature.
2019-03-07 12:41:40 +01:00
Antonio Ospite
38285e5bcf rtpsession: fix typo in a comment, s/SESSION_LOCK/RTP_SESSION_LOCK/
Fix a typo in a comment, mainly to avoid confusing autocompletion in
text editors.
2019-03-07 12:41:40 +01:00
Antonio Ospite
43e4226844 rtpsession: fix typos and update parameters names in comments
Some functions now accept a generic 'gpointer data' parameter because
they can work either on a single buffer or a buffer list.

However the comments were still referring to the old 'GstBuffer *buffer'
parameter, so update the comments to match the actual functions
signature.
2019-03-07 12:41:40 +01:00
Antonio Ospite
b2b60c4d8f rtpstats: fix some fields names in the RTPSourceStats documentation
Fix documentation of RTPSourceStats to use the actual fields names.
2019-03-07 10:36:11 +01:00
Marc Leeman
8737e29a49 rtpsource: small spell correct 2019-02-27 16:14:22 +01:00
Nicolas Dufresne
e72ef633a6 rtpsession: Fix EOS forwarding
So far we assumed that if all sources are bye, this meant we needed to
send an EOS on the RTCP sink. The problem is that this case may happens
if we only had one internal source and it detected a collision.

So now we limit the EOS forwarding to when there is a send_rtp_sink pad
and that this pad has received EOS. We don'tcheck the recv_rtp_sink
since the code does not wait for the bye to be send before sending EOS
to the RTCP src pad.
2019-02-25 17:06:50 +00:00
Nicolas Dufresne
06c340edd4 rtp: Add property to disable RTCP reports per internal rtpsource
This is useful when implementing custom retransmission mechanism like
RIST to prevent RTCP from being produces for the retransmitted SSRC.
This would also be used in general for various purpose when customizing
an RTP base pipeline.
2019-02-13 15:07:39 -05:00
Olivier Crête
b88a3abf46 rtpsession: Emit on-new-sender-ssrc for RTX ssrc also 2019-02-13 15:07:39 -05:00
Olivier Crête
bf00ee46de rtpjitterbuffer: Limit size to 2^15 packets
If it goes over 2^15 packets, it will think it has rolled over
and start dropping all packets. So make sure the seqnum distance is not too big.

But let's not limit it to a number that is too small to avoid emptying it
needlessly if there is a spurious huge sequence number, let's allow at
least 10k packets in any case.
2019-02-11 23:41:14 +00:00
Olivier Crête
086bad4643 rtpjitterbuffer: There is no automatic reorder threshold 2019-02-11 11:33:36 -05:00
Mathieu Duponchelle
a6d681ad09 rtpjitterbuffer: use the correct segment seqnum 2019-02-04 13:14:37 +00:00
Mathieu Duponchelle
5e92f7d208 rtpsession: use the correct segment seqnum 2019-02-04 13:14:37 +00:00
Mathieu Duponchelle
f52e16ceb8 Revert "rtpbin: receive bundle support"
This reverts commit dcd3ce9751.

This functionality was implemented for gstopenwebrtc, but it
turned out this was not actually needed for webrtc bundling
support, as shown in webrtcbin. It also doesn't correspond
to any standards.

This is an API break, but nothing should actually depend on
this, at least not for its initial purpose.

Changes in rtpbin.c were reverted manually, to preserve some
refactoring that had occurred in the original commit.

Fixes #537
2018-12-20 13:25:10 +00:00
Olivier Crête
d857522237 rtpjitterbuffer: Run all timers immediately on EOS
When the EOS event is received, run all timers immediately and avoid
pushing the EOS downstream before this has been run. This ensures that
the lost packet statistics are accurate.
2018-12-14 12:10:16 +00:00
Nicolas Dufresne
3de2c28fc1 rtpjitterbuffer: Stop waiting after EOS
After EOS is received, it is pointless to wait for further events,
specially waiting on timers. This patches fixes two cases where we could
wait instead of returning GST_FLOW_EOS and trigger a spin of the loop
function when EOS is queued, regardless if this EOS is the queue head or
not.
2018-12-14 12:10:16 +00:00
Miguel Paris
48a4fd4e50 rtpsession: properly handle rtcp_feedback_retention_window
- Consider GST_CLOCK_TIME_NONE as not to be used.
- Complete "rtcp-feedback-retention-window" property getter/setter
  implementation.
2018-11-30 10:55:26 +00:00
Miguel Paris
458741e4b2 rtpsource: properly prune RTCP packets out of feedback_retention_window
Closes #522
2018-11-30 10:55:26 +00:00
Miguel Paris
53f03d4cc1 rtpsource: properly compare buffer PTSs 2018-11-30 10:55:26 +00:00
Miguel Paris
57829c3352 rtpsource: retain_rtcp_packet: warning if invalid running_time 2018-11-30 10:55:26 +00:00
Miguel Paris
36f55b03e8 rtpsession: properly set the running_time for rtcp packet info 2018-11-30 10:55:26 +00:00
Nicolas Dufresne
d637567ab3 rtpssrcdemux: Rename confusingly name lock macros
This is an extra internal recurisve lock use to avoid having to take
both sink pad streams lock all the time. This patch renamed it
INTERLNAL_STREAM_LOCK/UNLOCK() to avoid confusion with possible upstream
GST_PAD API.
2018-11-29 15:34:47 -05:00
Nicolas Dufresne
40daf6322d rtpssrcdemux: Hold on internal stream lock while pushing sticky
This reverts "6f3734c305 rtpssrcdemux: Only forward stick events while
holding the sinkpad stream lock" and actually hold on the internal
stream lock. This prevents in some needed case having a second
streaming thread poping in and messing up event ordering.
2018-11-29 15:33:57 -05:00
Jordan Petridis
515ada7e22
Run gst-indent through the files
This is required before we enabled an indent test in the CI.

https://gitlab.freedesktop.org/gstreamer/gstreamer-project/issues/33
2018-11-28 05:52:16 +02:00
Nicolas Dufresne
21378d83c2 rtpssrcdemux: Forward serialized events to all pads
While forwarding serialized event, we use gst_pad_forward() function.
In the forward callback (GstPadForwardFunction) we always return
TRUE. Returning true there will stop the dispatching procedure. As a
side effect, only one events is receiving the events. This breaks
when sending EOS from the applicaiton, it also breaks the latency
tracer.
2018-11-24 13:01:25 +00:00
Linus Svensson
8fc8b7ee33 rtpsession: Implement reset
Reset RTPSession when rtpsession changes state from PAUSED to READY.
Without this change, a stored last_rtptime in RTPSource could interfere
with RTP timestamp generation in RTCP Sender Report.

Fixes #510
2018-11-13 12:30:35 +00:00
Mathieu Duponchelle
fd560bcb27 rtpfunnel: Stop using G_DECLARE_FINAL_TYPE
Fixes #516
2018-11-13 00:37:11 +01:00
Sebastian Dröge
87202cc03d rtpbin: Sink jitterbuffer/storage before passing as parameters to signals
Otherwise signal handlers from bindings will take ownership of them as
they are still floating, and we won't own a reference inside rtpbin
anymore.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/515
2018-11-07 09:11:16 +00:00
Olivier Crête
cc69c876fe rtpsession: Allow changing the SDES at runtime
Make it possible to modify the SDES in a packet at runtime.

https://bugzilla.gnome.org/show_bug.cgi?id=763502
2018-10-28 12:10:36 +00:00
Mathieu Duponchelle
ee461fb326 rtpfunnel: fix shutdown
By disposing of the ssrc_to_pad map in finalize instead of
dispose.
2018-10-15 14:20:58 +02:00
Havard Graff
53a45b1222 Initial commit of GstRtpFunnel
For funneling together rtp-streams into a single session.
Use-cases include multiplexing and bundle.
2018-10-15 14:20:58 +02:00
Havard Graff
6c05180dc5 rtpmux: respect downstream "timestamp-offset" in caps.
https://bugzilla.gnome.org/show_bug.cgi?id=795162
2018-10-10 15:39:02 -04:00
Havard Graff
6f37bd8f19 rtpmux: cleanup ssrc-handling code a bit
And add some better logging.

https://bugzilla.gnome.org/show_bug.cgi?id=795162
2018-10-10 15:38:57 -04:00
Havard Graff
18a1dc4ab6 rtpmux: protect against NULL caps
Due to state-changes deactivating the pad from another thread,
this can happen.

https://bugzilla.gnome.org/show_bug.cgi?id=795162
2018-10-10 15:35:31 -04:00
Havard Graff
7cd36d2914 rtpmux: property should overrule both upstream and downstream
https://bugzilla.gnome.org/show_bug.cgi?id=762213

https://bugzilla.gnome.org/show_bug.cgi?id=795162
2018-10-10 15:35:31 -04:00
Havard Graff
ac6e77acad rtpsession: Don't start the RTCP thread until it's needed
Always wait with starting the RTCP thread until either a RTP or RTCP
packet is sent or received. Special handling is needed to make sure the
RTCP thread is started when requesting an early RTCP packet.

We want to wait with starting the RTCP thread until it's needed in order
to not send RTCP packets for an inactive source.

https://bugzilla.gnome.org/show_bug.cgi?id=795139
2018-07-12 18:37:33 +02:00
Tim-Philipp Müller
238a37295c Update for g_type_class_add_private() deprecation in recent GLib
https://gitlab.gnome.org/GNOME/glib/merge_requests/7
2018-06-23 23:44:19 +02:00
Tim-Philipp Müller
db688c5504 docs: fix typos 2018-05-23 13:14:27 +01:00
Havard Graff
4d54673cb4 rtpsession: make "clear-pt-map" action signal actually work
Needed for PLI + FIR unit tests in follow-up commit.

https://bugzilla.gnome.org/show_bug.cgi?id=795139
2018-05-15 11:52:14 +01:00
Mikhail Fludkov
40eb462591 rtpsession: Avoid unnecessary copy of stats structure
The code before copied GstStructure twice. The first time inside
gst_value_set_structure and the second time in g_value_array_append.
Optimized version does no copies, just transfers ownership to
GValueArray. It takes advantage of the fact that array has already
enough elements preallocated and the memory is zero initialized.

https://bugzilla.gnome.org/show_bug.cgi?id=795139
2018-05-15 11:33:01 +01:00
Stian Selnes
457fdf95c4 rtpsession: Drop packet if trying to send from non-internal source
If obtain_internal_source() returns a source that is not internal it
means there exists a non-internal source with the same ssrc. Such an
ssrc collision should be handled by sending a GstRTPCollision event
upstream and choose a new ssrc, but for now we simply drop the packet.
Trying to process the packet further will cause it to be pushed
usptream (!) since the source is not internal (see source_push_rtp()).

https://bugzilla.gnome.org/show_bug.cgi?id=795139
2018-05-15 10:34:29 +01:00
Havard Graff
b43ee8f5b1 rtpsession: Try media_ssrc if no src can be found for PLI sender_ssrc
Some RTP stacks out there does not set the sender_ssrc. In order to be
more robust, try to lookup the media_ssrc before dropping the PLI.

https://bugzilla.gnome.org/show_bug.cgi?id=795139
2018-05-13 20:41:39 +01:00
Mikhail Fludkov
386ca1d378 rtpsession: Fix on-feedback-rtcp race
If there is an external source which is about to timeout and be removed
from the source hashtable and we receive feedback RTCP packet with the
media ssrc of the source, we unlock the session in
rtp_session_process_feedback before emitting 'on-feedback-rtcp' signal
allowing rtcp timer to kick in and grab the lock. It will get rid of
the source and rtp_session_process_feedback will be left with RTPSource
with ref count 0.

The fix is to grab the ref to the RTPSource object in
rtp_session_process_feedback.

https://bugzilla.gnome.org/show_bug.cgi?id=795139
2018-05-13 20:33:56 +01:00
Stian Selnes
29f26e8768 rtpsession: Add missing lock around sess->ssrcs iteration
https://bugzilla.gnome.org/show_bug.cgi?id=795139
2018-05-13 19:17:02 +01:00
John-Mark Bell
0a2b55ac3c rtpsession: do not emit RBs for internal senders.
These are the sources we send from, so there is no reason to
report receive statistics for them (as we do not receive on them,
and the remote side has no knowledge of them).

https://bugzilla.gnome.org/show_bug.cgi?id=795139
2018-05-13 19:16:59 +01:00
Xavier Claessens
edd9c8f6b8 Meson: Generate pc file for all plugins in good
https://bugzilla.gnome.org/show_bug.cgi?id=794568
2018-04-25 11:07:06 +01:00
Mathieu Duponchelle
8270cbacb4 rtxsend: fix wrong memory layout assumption
The code responsible for creating retransmitted buffers
assumed the stored buffer had been created with
rtp_buffer_new_allocate when copying the extension data,
which isn't necessarily the case, for example when
the rtp buffers come from a udpsrc.

https://bugzilla.gnome.org/show_bug.cgi?id=794958
2018-04-06 20:25:04 +02:00
Mathieu Duponchelle
4f8b34ab85 rtpbin: new signal "get-storage"
Similar to the get-session and get-internal-session signals,
we expose a get-storage signal in addition to the
get-internal-storage signal to give access to the actual
element for applications that need to set properties on the
element, in particular "size-time"

https://bugzilla.gnome.org/show_bug.cgi?id=794910
2018-04-06 20:21:43 +02:00
Tim-Philipp Müller
b387989bc6 docs: rtpbin: add some Since markers for new properties 2018-03-12 13:21:08 +00:00
Mathieu Duponchelle
d2b51bd727 rtpptdemux: provide example usage for ignored-payload-types 2018-02-26 17:02:52 +01:00
Mathieu Duponchelle
55ecde7ee5 rtpbin, rtpptdemux: Add missing Since markers 2018-02-26 16:53:08 +01:00
Mathieu Duponchelle
359b0a86f1 rtpptdemux: do no assume sink caps are non NULL 2018-02-21 19:59:04 +01:00
Mathieu Duponchelle
fdf64195ac rtpbin: Expose FEC support signals
Also slightly refactor complete_session_src

https://bugzilla.gnome.org/show_bug.cgi?id=792696
2018-02-21 14:15:22 +01:00
Mathieu Duponchelle
82d0950254 rtpptdemux: Add ignored-payload-types property
Packets with these payload types will be dropped. A use case
for this is FEC, where we want FEC packets to go through the
jitterbuffer, but not be output by rtpbin.

https://bugzilla.gnome.org/show_bug.cgi?id=792696
2018-02-21 14:15:22 +01:00
Mathieu Duponchelle
36b991f0b3 rtpptdemux: Add ssrc to output caps
It may be useful downstream

https://bugzilla.gnome.org/show_bug.cgi?id=792696
2018-02-21 14:15:22 +01:00
Patrick Radizi
364dbb5fc7 rtpjitterbuffer: allow timestamps to move backwards
The original solution for #784002 incorrectly assumed that timestamps
may not move backwards and changed timestamps that did so.

https://bugzilla.gnome.org/show_bug.cgi?id=784002
2018-02-15 10:05:39 +02:00
Mathieu Duponchelle
03dc22951b rtpbin: fix leak of elements requested by signals
When the signal returns a floating reference, as its return type
is transfer full, we need to sink it ourselves before passing
it to gst_bin_add (which is transfer floating).

This allows us to unref it in bin_remove_element later on, and
thus to also release the reference we now own if the signal
returns a non-floating reference as well.

As we now still hold a reference to the element when removing it,
we also need to lock its state and setting it to NULL before
unreffing it

Also update the request_aux_sender test.

https://bugzilla.gnome.org/show_bug.cgi?id=792543
2018-01-18 15:26:43 +01:00
Haakon Sporsheim
3c0d006c03 rtpsession: Handle zero length feedback packets
https://bugzilla.gnome.org/show_bug.cgi?id=791074
2017-12-02 13:58:34 +00:00
Justin Kim
2a5aafe425 rtpsesson: downgrade message level to debug when detected XR
When XR packet is detected, warning message leads to misunderstandings.
Until RFC3611 is implemented in gst-plugins-base, the level needs to
be downgraded to avoid confusion.

https://bugzilla.gnome.org/show_bug.cgi?id=789746
2017-11-01 10:57:00 +02:00
Tim-Philipp Müller
6cb51bd8cf rtpjitterbuffer: fix debug message on pt mismatch 2017-10-08 00:07:43 +01:00
Tim-Philipp Müller
d5f72418c8 rtpbin, rtspsrc: fix compiler warnings about 64-bit integer signednes
"warning: this decimal constant is unsigned only in ISO C90" with
gcc 4.8.4 (Ubuntu/Linaro 4.8.4-2ubuntu1~14.04.3)
2017-10-07 15:55:24 +01:00
Tim-Philipp Müller
a802f5df42 rtpjitterbuffer: implement basic chain_list function
Doesn't do anything fancy yet, but still avoids lots of
unnecessary locking/unlocking that would happen if the
default chain_list fallback function in GstPad got invoked.
2017-09-17 16:33:15 +01:00
Patrick Radizi
3de0244532 rtpbin: add option for sanity checking timestamp offset
Timestamp offsets needs to be checked to detect unrealistic values
caused for example by NTP clocks not in sync. The new parameter
max-ts-offset lets the user decide an upper offset limit. There
are two different cases for checking the offset based on if
ntp-sync is used or not:
1) ntp-sync enabled
   Only negative offsest are allowed since a positive offset would
   mean that the sender and receiver clocks are not in sync.
   Default vaule of max-ts-offset = 0 (disabled)
2) ntp-sync disabled
   Both positive and negative offsets are allowed.
   Default vaule of max-ts-offset = 3000000000
The reason for different default values is to be backwards
compatible.

https://bugzilla.gnome.org/show_bug.cgi?id=785733
2017-09-15 13:33:14 +03:00
Patrick Radizi
23f7739ba4 rtpbin: add option for increasing ts_offset gradually
Instant large changes to ts_offset may cause timestamps to move
backwards and also cause visible effects in media playback. The new
option max-ts-offset-adjustment lets the application control the rate to
apply changes to ts_offset.

https://bugzilla.gnome.org/show_bug.cgi?id=784002
2017-09-14 13:15:56 +03:00
George Kiagiadakis
286e1e62be rtprtx{send,receive}: improve the debug messages
* use INFO/DEBUG/LOG/TRACE equaly and meaningfully;
  previously rtprtxsend:LOG and rtprtxreceive:LOG would generate
  a totally different amount of log traffic and sometimes it was
  impossible to see the information you wanted without useless
  spam being printed around
* improve the wording, give a reasonable and self-explanatory
  amount of information
* print SSRCs in hex
* avoid G_FOO_FORMAT for readability (we are just printing integers)
2017-09-07 14:43:32 +03:00
Sebastian Dröge
71104f452e rtpbin: Also log local and SR RTP running times when doing ntp-sync=true 2017-08-29 19:14:25 +03:00
Matthew Waters
f602b8e5b0 rtpbin: also create session when creating the send_rtcp_src_%u pad
If one requests the send_rtcp_src_%u pad before a recv_rtcp_sink_%u pad,
the session/pad would never be created and NULL was returned.
Switching the request order would work.

https://bugzilla.gnome.org/show_bug.cgi?id=786718
2017-08-29 12:47:30 +10:00
Mathieu Duponchelle
5e48e85fb7 rtpstats: fix unsigned integer comparisons.
Callers of the API (rtpsource, rtpjitterbuffer) pass clock_rate
as a signed integer, and the comparison "<= 0" is used against
it, leading me to think the intention was to have the field
be typed as gint32, not guint32.

This led to situations where we could call scale_int with
a MAX_UINT32 (-1) guint32 as the denom, thus raising an
assertion.

https://bugzilla.gnome.org/show_bug.cgi?id=785991
2017-08-11 13:29:24 +02:00
Olivier Crête
96e71b0286 rtpsession: Send EOS if all internal sources sent bye
The ones which are not internal should not matter, and we should
wait for all sources to have sent their BYEs.

And add unit test

https://bugzilla.gnome.org/show_bug.cgi?id=773218
2017-07-04 21:14:10 -04:00
Olivier Crête
7e7e52caa0 rtpsession: Only send EOS if all sources have been marked bye
Now that multiple sender RTPSource can share the same RTPSession, we
must not send an EOS unless they're all marked bye.
2017-07-04 13:36:44 -04:00
Nicolas Dufresne
bf5cbce3b4 rtprtxreceive: Add memory and boudary checks
This element was not checking if mapping the RTP buffer and the payload
worked, and was not checking if the RTX payload was large enough.

https://bugzilla.gnome.org/show_bug.cgi?id=784484
2017-07-04 09:58:15 -04:00
Julien Isorce
afbabaefbe rtpstats: fix assertion 'denom > 0' failed
gst_util_uint64_scale_int takes a gint as denom parameter
whereas ctx->clock_rate is a guint32.

It happens when gst_rtp_packet_rate_ctx_reset set clock_rate
to -1.

So just define clock_rate as gint like it is done in rtpsource.h

https://bugzilla.gnome.org/show_bug.cgi?id=784250
2017-06-29 15:58:44 -04:00
Nicolas Dufresne
bbe0053f8a rtpjitterbuffer: Add a faststart-min-packets property
When set this property will allow the jitterbuffer to start delivering
packets as soon as N most recent packets have consecutive seqnum. A
faststart-min-packets of zero disables this feature. This heuristic is
also used in rtpsource which implements the probation mechanism and a
similar heuristic is used to handle long gaps.

https://bugzilla.gnome.org/show_bug.cgi?id=769536
2017-06-28 11:51:10 -04:00
Juan Navarro
72d2afda18 rtpsession: print value of unknown RTCP Payload Type
This adds printing the actual value of any unknown RTCP PT
to the already existing WARNING log message.

https://bugzilla.gnome.org/show_bug.cgi?id=783248
2017-05-31 10:20:27 +03:00
Nicolas Dufresne
b68d936ae0 Remove plugin specific static build option
Static and dynamic plugins now have the same interface. The standard
--enable-static/--enable-shared toggle are sufficient.
2017-05-16 14:41:19 -04:00
Tim-Philipp Müller
50a4b5bc0d Revert "rtpbin: pipeline gets an EOS when any rtpsources byes"
This reverts commit eeea2a7fe8.

It breaks EOS in some sender pipelines, see
https://bugzilla.gnome.org/show_bug.cgi?id=773218#c20
2017-04-19 12:28:12 +01:00
George Kiagiadakis
7f6c783930 rtprtxqueue: implement handling of the max-size-time property
https://bugzilla.gnome.org/show_bug.cgi?id=780867
2017-04-11 09:44:33 +03:00
George Kiagiadakis
501bf0e8d1 rtpmux: fix output segment and buffer DTS to correspond to the flattened PTS
https://bugzilla.gnome.org/show_bug.cgi?id=780347
2017-03-24 11:09:46 +02:00
George Kiagiadakis
3e91601fbb rtprtxqueue: add basic documentation and example pipelines
Mostly explaining the difference between rtprtxqueue and rtprtxsend.
2017-03-20 12:10:55 +02:00
George Kiagiadakis
ba606b96d3 rtprtxreceive: fix example pipelines and improve the documentation
https://bugzilla.gnome.org/show_bug.cgi?id=771383
2017-03-17 19:07:34 +02:00
George Kiagiadakis
0e65304d5c rtpsession: print the correct variable in debug statement
This debug statement is meant to print the time since the last (early)
RTCP transmission, not the last regular RTCP transmission (which also
happens to be set a few lines above to current_time, so the debug output
is just confusing)
2017-03-16 17:46:46 +02:00