Commit graph

92 commits

Author SHA1 Message Date
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
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
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
Tim-Philipp Müller
d7b2820b73 Fix indentation 2017-01-09 19:05:10 +00:00
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
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
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
Jan Schmidt
a660ac7e88 rtpjitterbuffer: Fix debug output when resyncing
Don't output the pointer value of the time() function as a timestamp
by using the correct variable.

Fixes build on Raspberry Pi 3.
2016-04-15 14:35:07 +00:00
Sebastian Dröge
4a0de53cc1 rtpjitterbuffer: Fix rtp_jitter_buffer_get_ts_diff() fill level calculation
The head of the queue is the oldest packet (as in lowest seqnum), the tail is
the newest packet. To calculate the fill level, we should calculate tail-head
while considering wraparounds. Not the other way around.

Other code is already doing this in the correct order.

https://bugzilla.gnome.org/show_bug.cgi?id=764889
2016-04-12 10:17:57 +03:00
Sebastian Dröge
df247f091c rtpjitterbuffer: Add RFC7273 media clock handling
https://bugzilla.gnome.org/show_bug.cgi?id=762259
2016-04-03 11:24:34 +03:00
Sebastian Dröge
366bbffcd8 Revert "WIP: rtpjitterbuffer: Add RFC7273 media clock handling"
This reverts commit 271501f657.

It wasn't meant to be pushed yet as the commit message indicates.
2016-01-18 11:30:45 +02:00
Sebastian Dröge
271501f657 WIP: rtpjitterbuffer: Add RFC7273 media clock handling 2016-01-18 08:58:59 +02:00
Sangkyu Park
2663388000 rtpjitterbuffer: Minor clean-up
1. Fix the code which is wrong coding style.
2. Fix a typing error of comment.

https://bugzilla.gnome.org/show_bug.cgi?id=751316
2015-06-22 13:08:12 +02:00
Sangkyu Park
6696bd62ef rtpjitterbuffer: Minor cleanup
1. Add Null check in 'free_item' function.
2. Fix a typing error of comment.

https://bugzilla.gnome.org/show_bug.cgi?id=750965
2015-06-15 11:55:57 +02:00
Sebastian Dröge
6c27293ffe rtpjitterbuffer: Change resyncing GST_WARNING to GST_INFO
This also happens in the very beginning when we receive the first packet, a
warning would be very confusing here. In all places where we should warn about
this, we would've printed a warning already before.
2015-04-13 20:25:48 +02:00
Wim Taymans
2e7f5c08cf jitterbuffer: rework resync handling
Add a need-resync state, this is when we need to try to lock on to a
time/RTPtime pair.
Always check the RTP timestamps and if they go backwards, mark ourselves
as need-resync.
Only resync when need-resync is TRUE and we have a valid time. Otherwise
we keep the old values. This avoids locking on to an invalid time and
causing us to timestamp everything with -1.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730417
2014-10-21 11:57:34 +02:00
Stefan Sauer
98222a67ff rtpjitterbuffer: don't log all clock_rate changes as warnings.
We never initialize clock_rate explicitly, therefore it is 0 by default. The
parameter is a uint32 and the only caller ensure that it is >0, therefore it
won't become -1 ever.
2014-10-04 17:17:13 +02:00
Aleix Conchillo Flaqué
bcd469ff31 rtpjitterbuffer: don't stop looping if event found in the queue
If we are inserting a packet into the jitter queue we need to keep
looping through the items until the right position is found. Currently,
the code stops as soon as an event is found in the queue.

Regarding events, we should only move packets before an event if there
is another packet before the event that has a larger seqnum.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730078
2014-05-14 10:23:28 +02:00
Wim Taymans
18b69419fd rtpjitterbuffer: rework packet insert
Rework the packet queue so that the most common action (insert a packet
at the tail of the queue) goes very fast.

Report if a packet was inserted at the head instead of the tail so that
we can know when to retry _pop or _peek.
2014-04-29 16:02:37 +02:00
Wim Taymans
3e11ce43b9 jitterbuffer: improve EOS handling
Make a new method to disable the jitterbuffer buffering.
Rework the update_estimated_eos() method. Calculate how much time
there is left to play. If we have less than the delay of the
jitterbuffer, we disabled buffering because we might never be able to
fill the complete jitterbuffer again.
If we receive an EOS event, disable buffering. We will drain the
buffer and eventually push the EOS event out.
When we reach the estimated NPT timeout and we didn't receive an EOS
event, make one and queue it so that it can be pushed.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728017
2014-04-18 14:07:31 +02:00
Wim Taymans
b4caf09011 jitterbuffer: fix buffer level with invalid DTS
It is possible that the DTS is invalid (when we receive RTP packets from
TCP, for example). As a fallback, use the reconstructed PTS value to
calculate the buffer level.
2014-03-03 11:34:00 +01:00
Wim Taymans
4a2e0f4ff4 rtpjitterbuffer: reorganize jitterbuffer items
Keep the oldest item at the head and the newest items on the tail. This
makes it easier to deal with -1 seqnums.
2013-12-10 11:01:03 +01:00
Wim Taymans
ea2a222cef jitterbuffer: correctly check for invalid values
Check for -1 on the guint from the buffer item instead of on the guint16
or guint32.
Also insert -1 seqnum at the head of the jitterbuffer.
2013-12-09 23:34:10 +01:00
Wim Taymans
0c6f4efe4a rtpjitterbuffer: avoid mapping the buffer
Reuse the parsed structure to get the timestamps.
2013-11-19 10:12:00 +01:00
Wim Taymans
43645d5981 jitterbuffer: add new timestamp mode
Add a new timestamp mode that assumes the local and remote clock are
synchronized. It takes the first timestamp as a base time and then uses the RTP
timestamps for the output PTS.
2013-10-31 10:15:25 +01:00
Wim Taymans
6e7d547be4 rtpjitterbuffer: reset skew does not reset clock-rate
Don't reset the clock-rate when we reset the skew correction algorithm.
Reset the skew correction algorithm when we change the clock-rate.
2013-09-30 11:53:08 +02:00
Wim Taymans
d4b4b4e924 rtpjitterbuffer: don't calculate skew without rtptime
Skip trying to calculate the skew when we don't have an rtptime.
It causes problems when lost packet events are placed in the jitterbuffer.
2013-09-26 16:21:33 +02:00
Wim Taymans
ae389aeb0c rtpjitterbuffer: remove list debug 2013-09-23 14:45:26 +02:00
Wim Taymans
479c7642fd rtpjitterbuffer: fix flush
Pass function to flush to properly free the queue items.
2013-09-23 14:45:25 +02:00
Wim Taymans
0cc887eb98 rtpjitterbuffer: append seqnum -1 packets 2013-09-23 14:45:25 +02:00
Wim Taymans
39a2ba669d rtpjitterbuffer: use structure to hold packet information
Make the jitterbuffer operate on a structure containing all the packet
information. This avoids mapping the buffer multiple times just to get the RTP
information. It will also make it possible to store other miniobjects such as
events later.
2013-09-23 14:45:25 +02:00
Wim Taymans
5051f51f0a jitterbuffer: configure clock-rate on jitterbuffer
Add a get and setter to configure the clock-rate in the jitterbuffer instead of
passing it as an argument to the insert method.
2013-09-23 14:45:24 +02:00
Wim Taymans
f864131227 jitterbuffer: operate on DTS
Make the jitterbuffer schedule the timeouts based on the DTS instead
of the PTS. This makes it all smoother with reordered frames and gives
the decoder time to reorder the frames in time.
2013-08-01 11:36:56 +02:00
Wim Taymans
ea931d4f57 rtpjitterbuffer: use NULL to ignore percent
If we pass NULL to pop and push we ignore the percent result.
2013-07-30 23:24:23 +02:00
Markovtsev Vadim
7cebe2fc41 rtpjitterbuffer: improve debug output
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688935
2013-02-07 14:32:26 +01:00
Wim Taymans
c3077012c0 jitterbuffer: do skew estimation only for new timestamps
Only run the skew estimation code when we have a new RTP timestamp. If we have
the same RTP timestamp, we simply use the previous estimation. This works
because the new observation with the same RTP timestamp has to have a bigger
receiver time and is thus not going to influence the estimation except for
causing more jitter.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=640023
2013-02-06 17:15:11 +01:00
Tim-Philipp Müller
230cf41cc9 Fix FSF address
https://bugzilla.gnome.org/show_bug.cgi?id=687520
2012-11-04 00:07:18 +00:00
Wim Taymans
f4eef3f48d rtpbin: set PTS and DTS in jitterbufffer 2012-10-17 12:46:32 +02:00
Edward Hervey
86a57e3546 rtpmanager: Initialize GstRTPBuffer before usage 2011-12-05 18:40:12 +01:00
Wim Taymans
87fbd1e784 Merge branch 'master' into 0.11
Conflicts:
	common
	ext/pulse/pulsesink.c
	ext/soup/gstsouphttpclientsink.c
	gst/audioparsers/gstaacparse.c
	gst/audioparsers/gstac3parse.c
	gst/rtp/gstrtph264depay.c
	gst/rtpmanager/gstrtpjitterbuffer.c
	gst/rtpmanager/rtpjitterbuffer.c
	gst/rtsp/gstrtspsrc.c
	sys/ximage/gstximagesrc.c
2011-09-28 12:44:59 +02:00
Mark Nauwelaerts
915db26029 rtpjitterbuffer: only reset skew on gap if input ts available 2011-09-19 11:56:44 +02:00
Mark Nauwelaerts
1e17e10f75 rtpjitterbuffer: check some more for possible rtp timestamp discontinuity
... when operating in non slave mode, and reset if detected.
This should avoid some (large) bogus outgoing timestamp due to jumps
in rtp time, as result of PAUSE/PLAY or seek or ...
2011-09-19 11:56:40 +02:00
Mark Nauwelaerts
ef02634dc6 rtpmanager: port to 0.11
* use G_DEFINE_TYPE
* adjust to new GstBuffer and corresponding rtp and rtcp buffer interfaces
* misc caps and segment handling changes

FIXME: also relies on being able to pass caps along with a buffer,
which has no evident equivalent yet, so that either needs one,
or still needs quite some code path modification to drag along caps.
2011-07-06 10:16:12 +02:00
Miguel Angel Cabrera Moya
977a5eee7a rtpjitterbuffer: return correct type when assertion fails 2011-06-24 11:59:01 +02:00
Stefan Kost
9f34b89245 rtpjitterbuffer: don't divide by 0 2011-01-25 21:57:57 +02:00
Wim Taymans
eb6d552353 jitterbuffer: get better buffering level
When the jitterbuffer contains -1 timestamps, make sure we still calculate the
buffer fill level by skipping the -1 buffers.
Try to be more resilient to weird input timestamps.
2010-12-20 15:56:50 +01:00
Vladimir Eremeev
8bf7381385 rtpjitterbuffer: improve article reference in comment block
https://bugzilla.gnome.org/show_bug.cgi?id=631082
2010-10-01 18:07:03 +01:00
Thijs Vermeir
2c2c90a723 rtpjitterbuffer: update link to documentation 2010-09-30 12:08:49 +02:00
Wim Taymans
0f59664c6a rtpjitterbuffer: move comment where it belongs 2010-09-06 11:01:57 +02:00
Sebastian Dröge
0a12e69024 rtpjitterbuffer: Return 100 if high-level is 0 instead of dividing by zero 2010-02-23 17:20:02 +01:00