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1191 commits

Author SHA1 Message Date
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
Nirbheek Chauhan
b09f478e80 Add support for Meson as alternative/parallel build system
https://github.com/mesonbuild/meson

With contributions from:

Tim-Philipp Müller <tim@centricular.com>
Jussi Pakkanen <jpakkane@gmail.com> (original port)

Highlights of the features provided are:
* Faster builds on Linux (~40-50% faster)
* The ability to build with MSVC on Windows
* Generate Visual Studio project files
* Generate XCode project files
* Much faster builds on Windows (on-par with Linux)
* Seriously fast configure and building on embedded

... and many more. For more details see:

http://blog.nirbheek.in/2016/05/gstreamer-and-meson-new-hope.html
http://blog.nirbheek.in/2016/07/building-and-developing-gstreamer-using.html

Building with Meson should work on both Linux and Windows, but may
need a few more tweaks on other operating systems.
2016-08-20 11:21:12 +01:00
Thomas Bluemel
4dff74358e rtpjitterbuffer: Actually calculate the packet rate for max-dropout and max-misorder calculations.
https://bugzilla.gnome.org/show_bug.cgi?id=751311
2016-08-10 19:49:27 +02:00
Thomas Bluemel
e7d4ad7ac7 rtpjitterbuffer: Don't warn for duplicate packets
This is a normal scenario and should not be a warning.  This can
happen frequently when re-transmits of lost packets are enabled.

https://bugzilla.gnome.org/show_bug.cgi?id=762208
2016-08-10 19:39:42 +02:00
Thiago Santos
7f0381fdd9 rtpjitterbuffer: avoid unref of null buffer
The current 'l' pointer will be NULL when the loop
is interrupted with a 'break' statement. Need to have
it advance to the next list item before interrupting.
2016-08-04 00:36:28 -03:00
Aurélien Zanelli
f8f8935c77 rtpjitterbuffer: fix RTPJitterBufferMode documentation
Documentation lacks '@' before each enum values and there was an extra
line after symbol section which confuses GTK-Doc parser.

https://bugzilla.gnome.org/show_bug.cgi?id=767788
2016-06-17 15:16:45 +03:00
Miguel París Díaz
83f4c08747 rtpsession: take the lock when changing stats
https://bugzilla.gnome.org/show_bug.cgi?id=766025
2016-06-17 12:52:29 +03:00
Olivier Crête
5328378132 rtpjitterbuffer: Work with non-TIME segments
With non-time segments, it now assumes that the arrival time of packets
is not relevant and that only the RTP timestamp matter and it produces
an output segment start at running time 0.

https://bugzilla.gnome.org/show_bug.cgi?id=766438
2016-06-08 14:49:49 -04:00
Miguel París Díaz
389e0abeb0 rtpsource: complete warn log with SSRC
https://bugzilla.gnome.org/show_bug.cgi?id=767195
2016-06-06 10:47:17 +03:00
Mikhail Fludkov
ee7e80d615 rtpsession: don't act on suspicious BYE RTCP
Some endpoints (like Tandberg E20) can send BYE packet containing our
internal SSRC. I this case we would detect SSRC collision and get rid
of the source at some point. But because we are still sending packets
with that SSRC the source will be recreated immediately.
This brand new internal source will not have some variables incorrectly
set in its state. For example 'seqnum-base` and `clock-rate` values will be
-1.
The fix is not to act on BYE RTCP if it contains internal or unknown
SSRC.

https://bugzilla.gnome.org/show_bug.cgi?id=762219
2016-05-20 09:28:39 +03:00
Sebastian Dröge
fe34f46f32 rtpsession: Take the lock already when reading the other stats, not just for the hash table
https://bugzilla.gnome.org/show_bug.cgi?id=766025
2016-05-15 12:31:33 +03:00
Olivier Crête
0ebdb97797 jitterbuffer: Upgrade debug message to error
It causes the entire pipeline to fail, it should be easier to find.
2016-05-14 12:36:08 +02:00
Sebastian Dröge
204a86af97 rtpsession: Don't notify about stats property changes while taking the session lock
The signal handlers might want to actually get the value of the stats
property, which would take the session lock again and deadlock.

This was introduced by 2e960e7075.

https://bugzilla.gnome.org/show_bug.cgi?id=766025
2016-05-11 09:28:13 +03:00
Havard Graff
8f7962e1c3 rtpjitterbuffer: Fix stall when receiving already lost packet
When a packet arrives that has already been considered lost as part of a
large gap the "lost timer" for this will be cancelled. If the remaining
packets of this large gap never arrives, there will be missing entries
in the queue and the loop function will keep waiting for these packets
to arrive and never push another packet, effectively stalling the
pipeline.

The proposed fix conciders parts of a large gap definitely lost (since
they are calculated from latency) and ignores the late arrivals.

In practice the issue is rare since large gaps are scheduled immediately,
and for the stall to happen the late arrival needs to be processed
before this times out.

https://bugzilla.gnome.org/show_bug.cgi?id=765933
2016-05-06 14:32:42 +03:00
Miguel París Díaz
2e960e7075 rtpsession: Take session lock when creating stats
The access to the session hash table must happen while the session lock is
taken, otherwise another thread might modify the hash table while we're
creating the stats.

https://bugzilla.gnome.org/show_bug.cgi?id=766025
2016-05-06 09:24:22 +03:00