Commit graph

142 commits

Author SHA1 Message Date
Havard Graff
97ced29277 rtpsource: rewrite timeout-check to avoid underflow
If current_time is < collision_timeout, we get an uint64 underflow, and
the check will trigger prematurely.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/817>
2020-11-13 21:30:06 +01:00
Olivier Crête
99723bc1c1 rtpsource: Report for which local SSRC is a remote RB reporting on
This is useful in the Bundle case because there may be multiple local
and remote SSRCs in the same session.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/776>
2020-11-03 12:35:54 -05:00
Sebastian Dröge
e4ce9887cd rtpmanager: Improve readability of "stats" docs by making the fields an actual list
Otherwise they end up all in the same line one after another.

Also add docs for the "avg-jitter" stats field of the jitterbuffer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/698>
2020-08-13 07:24:17 +00:00
Mathieu Duponchelle
aa34c29d3b rtpmanager: fix various documentation issues
Improper naming of properties, improper links, misc

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/684>
2020-07-27 13:51:15 +00:00
Havard Graff
1df706448c rtpmanager: Google Transport-Wide Congestion Control RTP Extension
Generating and parsing the RTCP-messages described in:
https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions-01
2020-02-14 10:09:02 +00:00
Olivier Crête
c00796eaa5 rtpsession: Add test for packet rate maths 2020-02-06 14:01:38 -05:00
Aaron Boxer
46989dca96 documentation: fix a number of typos 2019-10-05 22:38:11 +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
Nicolas Dufresne
a6e7f258ac rtpsource: Add more information to probation warning 2019-05-02 14:44:58 -04:00
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
Antonio Ospite
d6939c4031 docs: fix typo s/incomming/incoming/ 2019-04-03 16:38:56 +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
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
Marc Leeman
8737e29a49 rtpsource: small spell correct 2019-02-27 16:14:22 +01:00
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
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
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
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
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
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
Havard Graff
b33470f80c rtpsource: reordered packets are very normal, and should not be a warning 2016-08-26 11:53:22 -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
Nirbheek Chauhan
78847d03cf rtpmanager: Some comment and documentation clarifications/fixes 2016-03-15 09:32:47 +00:00
Luis de Bethencourt
d4f094f587 rtpmanager: use GST_STIME_ARGS for GstClockTimeDiff
No need to manually handle negative values of diff, GST_STIME_ARGS does
exactly this.
2015-11-03 14:26:32 +00:00
Miguel París Díaz
f321bfeaf4 rtpmanager: Take into account packet rate for max-dropout and max-misorder calculations
https://bugzilla.gnome.org/show_bug.cgi?id=751311
2015-10-07 12:07:18 +01:00
Miguel París Díaz
4c96094fbb rtpmanager: add "max-dropout-time" and "max-misorder-time" props
https://bugzilla.gnome.org/show_bug.cgi?id=751311
2015-10-07 12:06:47 +01:00
Hyunjun Ko
b814d7ed25 rtpsource: doesn't handle probation and rtp gap in case of sender
https://bugzilla.gnome.org/show_bug.cgi?id=754548
2015-10-02 16:42:36 +03:00
Stefan Sauer
12930c2f8c docs: fix "Symbol name not found at the start of the comment block"
Add symbols or change comment into a regular comment.
2015-07-07 17:12:02 +02:00
Jose Antonio Santos Cadenas
11f298a338 rtpsource: Do not try to push NULL buffers
If update_receiver_stats() fails, we can't really do anything with this buffer
anymore and have to drop it. This happens if there's a big seqnum
discontinuity for example.

https://bugzilla.gnome.org/show_bug.cgi?id=751311
2015-06-22 12:26:59 +02:00
Sebastian Dröge
dc059efa60 rtp: Use GST_BUFFER_PTS() instead of GST_BUFFER_TIMESTAMP()
The mix between all these in the RTP code is confusing, let's try to be
consistent.
2015-06-10 14:34:47 +02:00
Ilya Konstantinov
c7e168ec70 rtpmanager: clarify negative lost packets in stats
Also:
- Move notes on units before field documentation.
- Unify documentation style.

https://bugzilla.gnome.org/show_bug.cgi?id=750653
2015-06-10 14:10:52 +02:00
Ilya Konstantinov
0a578c235a rtpmanager: document units of stats and arguments
Also, minor spelling and style corrections.

https://bugzilla.gnome.org/show_bug.cgi?id=750653
2015-06-09 18:21:59 +02:00
Sebastian Dröge
c60038f188 rtpsource: Queue bad packets instead of dropping them
So we can send them out once we found the next, consecutive sequence number in
case one is following.
2015-05-18 18:43:16 +03:00
Sebastian Dröge
9f18a271f3 rtpsource: Use g_queue_foreach() to unref all buffers in queues 2015-05-18 18:43:16 +03:00
Sebastian Dröge
54e924332e rtpsource: Refactor seqnum comparison code a bit 2015-05-18 18:43:16 +03:00
Sebastian Dröge
1974b24ef4 rtpsource: Allow sequence number wraparound during probation 2015-05-18 18:43:16 +03:00
Sebastian Dröge
3386de7a8a rtpsource: Make sequence number comparison code more readable
... by using gst_rtp_buffer_compare_seqnum() and signed integers
instead of implictly using effects of integer over/underflows.
2015-05-18 18:43:16 +03:00
Luis de Bethencourt
671b4d25cd remove unused enum items PROP_LAST
This were probably added to the enums due to cargo cult programming and are
unused. Removing them.
2015-04-24 17:01:12 +01:00
Sebastian Dröge
80268e7d37 rtpsource/rtprtxsend: Also pass correct seqnum-offset and payload to the RTX rtpsource
https://bugzilla.gnome.org/show_bug.cgi?id=747394
2015-04-16 17:33:37 +02:00
Santiago Carot-Nemesio
e05378ec16 rtp: Add Full Intra Request (FIR) packets to statistics
https://bugzilla.gnome.org/show_bug.cgi?id=745587
2015-03-04 12:04:40 +01:00
Santiago Carot-Nemesio
22791413f9 rtp: Add Packet Loss Indication (PLI) to statistics
This is helpful to provide statistics in the format defined in
http://w3c.github.io/webrtc-stats/#dictionary-rtcrtpstreamstats-members.

https://bugzilla.gnome.org/show_bug.cgi?id=745587
2015-03-04 12:04:07 +01:00
Olivier Crête
51a8bedced rtpsource: Rename seqnum-base to seqnum-offset in caps
This was modified back in 1.0 in GstRtpBasePayload
2014-10-10 18:33:34 -04:00