Commit graph

1405 commits

Author SHA1 Message Date
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
George Kiagiadakis
1622d4c894 rtprtxsend: convert LOG message to TRACE
This is printed too often (for every chained buffer!) and just clutters the logs.
2017-03-16 17:46:46 +02:00
Miguel París Díaz
9ffef7ecd5 rtpsource: fix warning message
https://bugzilla.gnome.org/show_bug.cgi?id=780105
2017-03-16 16:33:02 +02:00
Miguel París Díaz
54a2f33e47 rtpsource: get clock-rate from pt if needed to generate SR
https://bugzilla.gnome.org/show_bug.cgi?id=780105
2017-03-16 15:48:37 +02:00
George Kiagiadakis
71b63d54fe rtprtxreceive: fix potential leak of old, unassociated, association requests
https://bugzilla.gnome.org/show_bug.cgi?id=722560
2017-03-01 10:50:43 +02:00
Andrew
76792a5c20 rtpjitterbuffer: Don't always reset PTS to 0 after a gap
In function rtp_jitter_buffer_calculate_pts: If gap in incoming RTP
timestamps is more than (3 * jbuf->clock_rate) we call
rtp_jitter_buffer_reset_skew which resets pts to 0. So components down
the pipeline (playes, mixers) just skip frames/samples until pts becomes
equal to pts before gap.

In version 1.10.2 and before this checking was bypassed for packets with
"estimated dts", and gaps were handled correctly.

https://bugzilla.gnome.org/show_bug.cgi?id=778341
2017-02-26 12:41:19 +02:00
Miguel París Díaz
3aa69ca0bb rtpsession: relate received FIRs and PLIs to source
This is needed in order to:
 - Avoid ignoring requests for different media sources.
 - Add SSRC field in the GstForceKeyUnit event.

https://bugzilla.gnome.org/show_bug.cgi?id=778013
2017-02-02 12:13:59 -05:00
Santiago Carot-Nemesio
a1e4249131 rtpstats: Keep number of nacks sent/received per source
Currently, the nack packets sent or received are kept at session level,
which makes it impossible to distinguish how many of these packages were
sent/received per ssrc when several sources are in the same session. This
patch is aligned with the https://www.w3.org/TR/webrtc-stats/#dom-rtcrtpstreamstats

https://bugzilla.gnome.org/show_bug.cgi?id=776714
2017-01-24 12:38:50 +02:00
Mathieu Duponchelle
191330cba8 rtxqueue: Expose basic statistics as properties.
Statistics about the total number of retransmission requests
and the actual number of retransmitted packets can be helpful
at application-level.

https://bugzilla.gnome.org/show_bug.cgi?id=777182
2017-01-12 19:49:30 +01:00
Tim-Philipp Müller
d7b2820b73 Fix indentation 2017-01-09 19:05:10 +00:00
Reynaldo H. Verdejo Pinochet
264be35e3c rtpmanager: place content before Since-version API marker
Avoids confusing the parser
2016-12-14 14:38:38 -08:00
Edward Hervey
e5158ca496 jitterbuffer: Don't leak duplicate items
When providing items with a seqnum, there is a (very small) probability
that an element with the same seqnum already exists. Don't forget
to free that item if it wasn't inserted.

And avoid returning undefined values when dealing with duplicate items
2016-12-02 09:01:57 +01:00
Edward Hervey
91f5b4eaa2 rtprtxsend: Update statistics before pushing
If an element queries the number of retransmission buffers pushed
*while* the push is still taking place (and before the object lock
is taken just after) it would end up with the wrong statistic
being reported.

Increment it just before the push, avoids races when getting statistics

https://bugzilla.gnome.org/show_bug.cgi?id=768723
2016-11-27 11:15:49 +01:00
Sebastian Dröge
34db78b645 rtpbin: Handle create_session() returning NULL in bundle code
CID 1394492.
2016-11-23 18:34:04 +02:00
Sebastian Dröge
15630db146 rtpmux: Mark pad as needing reconfiguration again if it failed
And return FLUSHING instead of NOT_NEGOTIATED on flushing pads.

https://bugzilla.gnome.org/show_bug.cgi?id=774623
2016-11-18 12:04:45 +02:00
Philippe Normand
dcd3ce9751 rtpbin: receive bundle support
A new signal named on-bundled-ssrc is provided and can be
used by the application to redirect a stream to a different
GstRtpSession or to keep the RTX stream grouped within the
GstRtpSession of the same media type.

https://bugzilla.gnome.org/show_bug.cgi?id=772740
2016-11-16 08:56:34 +01:00
Havard Graff
1a4393fb4d rtpjitterbuffer: fix timer-reuse bug
When doing rtx, the jitterbuffer will always add an rtx-timer for the next
sequence number.

In the case of the packet corresponding to that sequence number arriving,
that same timer will be reused, and simply moved on to wait for the
following sequence number etc.

Once an rtx-timer expires (after all retries), it will be rescheduled as
a lost-timer instead for the same sequence number.

Now, if this particular sequence-number now arrives (after the timer has
become a lost-timer), the reuse mechanism *should* now set a new
rtx-timer for the next sequence number, but the bug is that it does
not change the timer-type, and hence schedules a lost-timer for that
following sequence number, with the result that you will have a very
early lost-event for a packet that might still arrive, and you will
never be able to send any rtx for this packet.

Found by Erlend Graff - erlend@pexip.com

https://bugzilla.gnome.org/show_bug.cgi?id=773891
2016-11-04 16:56:56 +02:00
Havard Graff
fb9c75db36 rtpjitterbuffer: fix lost-event using dts instead of pts
The lost-event was using a different time-domain (dts) than the outgoing
buffers (pts). Given certain network-conditions these two would become
sufficiently different and the lost-event contained timestamp/duration
that was really wrong. As an example GstAudioDecoder could produce
a stream that jumps back and forth in time after receiving a lost-event.

The previous behavior calculated the pts (based on the rtptime) inside the
rtp_jitter_buffer_insert function, but now this functionality has been
refactored into a new function rtp_jitter_buffer_calculate_pts that is
called much earlier in the _chain function to make pts available to
various calculations that wrongly used dts previously
(like the lost-event).

There are however two calculations where using dts is the right thing to
do: calculating the receive-jitter and the rtx-round-trip-time, where the
arrival time of the buffer from the network is the right metric
(and is what dts in fact is today).

The patch also adds two tests regarding B-frames or the
“rtptime-going-backwards”-scenario, as there were some concerns that this
patch might break this behavior (which the tests shows it does not).
2016-11-04 16:51:20 +02:00
Havard Graff
bea35f97c8 rtpjitterbuffer: fix bug in reschedule_timer
The new timeout is always going to be (timeout + delay), however, the
old behavior compared the current timeout to just (timeout), basically
being (delay) off.

This would happen if rtx-delay == rtx-retry-timeout, with the result that
a second rtx attempt for any buffers would be scheduled immediately instead
of after rtx-delay ms.

Simply calculate (new_timeout = timeout + delay) and then use that instead.

https://bugzilla.gnome.org/show_bug.cgi?id=773905
2016-11-04 16:40:14 +02:00
Alejandro G. Castro
6e7816c589 rtpbin: avoid generating errors when rtcp messages are empty and check the queue is not empty
Add a check to verify all the output buffers were empty for the
session in a timout and log an error.

https://bugzilla.gnome.org/show_bug.cgi?id=773269
2016-11-01 20:17:20 +02:00
Alejandro G. Castro
eeea2a7fe8 rtpbin: pipeline gets an EOS when any rtpsources byes
Instead of sending EOS when a source byes we have to wait for
all the sources to be gone, which means they already sent BYE and
were removed from the session. We now handle the EOS in the rtcp
loop checking the amount of sources in the session.

https://bugzilla.gnome.org/show_bug.cgi?id=773218
2016-11-01 20:16:18 +02:00
Tim-Philipp Müller
cae9ec0ad8 ext, gst: fix indentation 2016-09-15 09:53:07 +01:00
Thomas Bluemel
567afdd4d3 rtpjitterbuffer: Fix calculating next_seqnum when dropping old buffers from a full queue.
Fixes calculating the next sequence number when a ITEM_TYPE_LOST with more than one
definitely lost packets is encountered.

https://bugzilla.gnome.org/show_bug.cgi?id=769757
2016-09-14 19:47:28 -04:00
Havard Graff
f440b074b1 rtpjitterbuffer: improved rtx-rtt averaging
The basic idea is this:
1. For *larger* rtx-rtt, weigh a new measurement as before
2. For *smaller* rtx-rtt, be a bit more conservative and weigh a bit less
3. For very large measurements, consider them "outliers"
   and count them a lot less

The idea being that reducing the rtx-rtt is much more harmful then
increasing it, since we don't want to be underestimating the rtt of the
network, and when using this number to estimate the latency you need for
you jitterbuffer, you would rather want it to be a bit larger then a bit
smaller, potentially losing rtx-packets. The "outlier-detector" is there
to prevent a single skewed measurement to affect the outcome too much.
On wireless networks, these are surprisingly common.

https://bugzilla.gnome.org/show_bug.cgi?id=769768
2016-09-14 19:37:50 -04:00
Stian Selnes
f8238f0a9f rtpjitterbuffer: Detect whether to assume equidistant spacing when loss
Assuming equidistant packet spacing when that's not true leads to more
loss than necessary in the case of reordering and jitter. Typically this
is true for video where one frame often consists of multiple packets
with the same rtp timestamp. In this case it's better to assume that the
missing packets have the same timestamp as the last received packet, so
that the scheduled lost timer does not time out too early causing the
packets to be considered lost even though they may arrive in time.

https://bugzilla.gnome.org/show_bug.cgi?id=769768
2016-09-14 19:37:50 -04:00
Stian Selnes
2eb7383816 rtpjitterbuffer: Don't request rtx if 'now' is past retry period
There is no need to schedule another EXPECTED timer if we're already
past the retry period. Under normal operation this won't happen, but if
there are more timers than the jitterbuffer is able to process in
real-time, scheduling more timers will just make the situation worse.
Instead, consider this packet as lost and move on. This scenario can
occur with high loss rate, low rtt and high configured latency.

https://bugzilla.gnome.org/show_bug.cgi?id=769768
2016-09-14 19:37:50 -04:00
Stian Selnes
ab49dfd0b2 rtpjitterbuffer: Fix lost duration when gap after lost timer
This patch fixes an issue with the estimated gap duration when there is
a gap immediately after a lost timer has been processed. Previously
there was a discrepancy beteen the gap in seqnum and gap in dts which
would cause wrong calculated duration. The issue would only be seen with
retranmission enabled since when it's disabled lost timers are only
created when a packet is received and the actual gap length and last dts
is known.

https://bugzilla.gnome.org/show_bug.cgi?id=769768
2016-09-14 19:37:50 -04:00
Havard Graff
dd020f5cc8 rtpjitterbuffer: Expose rtx-deadline as a property
The default -1 gives the old behavior.

https://bugzilla.gnome.org/show_bug.cgi?id=769768
2016-09-14 19:37:50 -04:00
Havard Graff
8087a8a31c rtpjitterbuffer: Improved expected-timer handling when gap > 0
https://bugzilla.gnome.org/show_bug.cgi?id=769768
2016-09-14 19:37:50 -04:00
Stian Selnes
38a7545003 rtpjitterbuffer: Major improvements for RTX stats
Stats should also be collected for unsuccessful packets.

rtx-rtt is very important for determining the necessary configured
latency on the jitterbuffer. It's especially important to be able to
increase the latency when retransmitted packets arrive too late and are
considered lost. This patch includes these late packets in the
calculation of the various rtx stats, making them more correct and
useful.

Also in the case where the original packet arrives after a NACK is sent,
the received RTX packet should update the stats since it provides useful
information about RTT.

The RTT is only updated if and only if all requested retranmissions are
received. That way the RTT is guaranteed to make sense. If not we don't
know which request the packet is a response to and the RTT may be bogus.
A consequence of this patch is that RTT is not updated for a request
when one of the RTX packets for that seqnum is lost, but that since
measured RTT will be more accurate.

The implementation store the RTX information from the timed out timers
and use this when the retransmitted packet arrives. For performance
these timers are stored separately from the "normal" timers in order to
not impact performance (see attached performance test).

https://bugzilla.gnome.org/show_bug.cgi?id=769768
2016-09-14 19:37:50 -04:00
Havard Graff
1b868cc9b1 rtpjitterbuffer: Add and expose more stats and increase testing of it
Add num-pushed and num-lost.
Expose num-late, num-duplicates and avg-jitter.

https://bugzilla.gnome.org/show_bug.cgi?id=769768
2016-09-14 19:37:50 -04:00
Stian Selnes
531199d5c4 rtxreceive: Set buffer flag for retransmitted packets
https://bugzilla.gnome.org/show_bug.cgi?id=769768
2016-09-14 19:37:50 -04:00
Havard Graff
1436fc01e9 rtpjitterbuffer: Option to disable rtx-delay-reorder
When disabled we can save some iterations over timers.

There is probably an argument for rtx-delay-reorder to exist, but
for normal operations, handling jitter (reordering) is something a
jitterbuffer should do, and this variable feels like functionality that
is not "in-sync" with what the jitterbuffer is trying to achieve.

Example: You have 50ms jitter on your network, and are receiving
audio packets with 10ms durations. An audio packet should not be
considered late until its rtx-timeout has expired (and hence a rtx-event
is sent), but with rtx-delay-reorder, events will be sent pretty much
all the time due to the jitter on the network.

Point being: The jitterbuffer should adapt its size to the measured network
jitter, and then rtx-delay-reorder needs to adapt as well, or simply
get out of the way and let the other (better) rtx-mechanisms do their job.

Also change find_timer to only use seqnum as an argument, since there
will only ever be one timer per seqnum at any given time. In the
one case where the type matters, the caller simply checks the type.

https://bugzilla.gnome.org/show_bug.cgi?id=769768
2016-09-14 19:37:50 -04:00
Olivier Crête
4fceb5050f Revert "rtpmux: fix PROP_TIMESTAMP_OFFSET range problems"
This broke API, so we need a better solution!

This reverts commit c7579d31a6.
2016-08-26 12:06:51 -04:00
Havard Graff
c7579d31a6 rtpmux: fix PROP_TIMESTAMP_OFFSET range problems
It could not set the offset for the full guint32 range.
2016-08-26 11:57:14 -04:00
Havard Graff
7ad7266163 rtpbin: introduce max-streams property
To be able to cap the number of allowed streams for one session.

This is useful for preventing DoS attacks, where a sender can change
SSRC for every buffer, effectively bringing rtpbin to a halt.

https://bugzilla.gnome.org/show_bug.cgi?id=770292
2016-08-26 11:57:06 -04:00
Havard Graff
b33470f80c rtpsource: reordered packets are very normal, and should not be a warning 2016-08-26 11:53:22 -04:00
Havard Graff
babc591707 rtpsession: degrade g_warning to GST_ERROR
So we don't blow up while investigating
2016-08-26 11:53:22 -04:00
Stian Selnes
61bc228a71 rtpsession: sanity check RTT before ignoring PLI/FIR 2016-08-25 18:28:44 -04:00
Stian Selnes
85a56f8ee3 rtpsession: handle sdes messages with non-utf8 more gracefully 2016-08-25 18:28:44 -04:00
Havard Graff
1ef896b29d gstrtpsession: refactor duplicate code into a function
Less code, easier to read, more consistent.

https://bugzilla.gnome.org/show_bug.cgi?id=770293
2016-08-23 15:09:03 -04:00
Vincent Penquerc'h
0fb0c0c8e6 rtpbin: fix typo in max-misorder-time property name 2016-08-23 17:19:17 +01:00