Commit graph

1114 commits

Author SHA1 Message Date
Jose Antonio Santos Cadenas
f563176349 rtpssrcdemux: Add support for reduce size rtcp
According to RFC 5506, reduce size packages can be sent, this
packages may not be compound, so we need to add support for
getting ssrc from other types of packages.

https://bugzilla.gnome.org/show_bug.cgi?id=750327
2015-06-05 10:30:15 +02:00
Jose Antonio Santos Cadenas
f8f23bbf5d rtpsession: Add support for receiving reduced size rtcp
See RFC 5506

https://bugzilla.gnome.org/show_bug.cgi?id=750332
2015-06-05 10:24:17 +02:00
Sebastian Dröge
647eefea67 rtpsession: Only schedule a timer when we actually have to send RTCP
Otherwise we will have 10s-100s of thread wakeups in feedback profiles, create
RTCP packets, etc. just to suppress them in 99% of the cases (i.e. if no
feedback is actually pending and no regular RTCP has to be sent).

This improves CPU usage and battery life quite a lot.

https://bugzilla.gnome.org/show_bug.cgi?id=746543
2015-06-02 11:38:15 +02:00
Sebastian Dröge
8ada98964d rtpsession: Remove useless goto
https://bugzilla.gnome.org/show_bug.cgi?id=746543
2015-06-02 11:38:15 +02:00
Sebastian Dröge
506a8a8857 rtpbin: Add rtp-profile property for setting the default profile of newly created sessions
https://bugzilla.gnome.org/show_bug.cgi?id=746543
2015-06-02 11:38:15 +02:00
Sebastian Dröge
0f7e80ed59 rtpsession: Only put RRs and full SDES into regular RTCP packets
If we may suppress the packet due to the rules of RFC4585 (i.e. when
below the t-rr-int), we can send a smaller RTCP packet without RRs
and full SDES. In theory we could even send a minimal RTCP packet
according to RFC5506, but we don't support that yet.

https://bugzilla.gnome.org/show_bug.cgi?id=746543
2015-06-02 11:38:15 +02:00
Sebastian Dröge
6f830e5bd5 rtpsession: Keep track of tp/tn and t_rr_last separately
Otherwise we can't properly schedule RTCP in feedback profiles as we need to
distinguish the time when we last checked for sending RTCP (tp) but might have
suppressed it, and the time when we last actually sent a non-early RTCP
packet.

This together with the other changes should now properly implement RTCP
scheduling according to RFC4585, and especially allow us to send feedback
packets a lot if needed but only send regular RTCP packets every once in a
while.

https://bugzilla.gnome.org/show_bug.cgi?id=746543
2015-06-02 11:38:15 +02:00
Sebastian Dröge
3122ef4ae3 rtpsession: Add property for selecting RTP profile (AVP/AVPF/etc)
And modify our RTCP scheduling algorithm accordingly. We now can send more
RTCP packets if needed for feedback, but will throttle full RTCP packets by
rtcp-min-interval (t-rr-int from RFC4585).

In non-feedback mode, rtcp-min-interval is Tmin from RFC3550, which is
statically set to 1s or 0s by RFC4585. Tmin defines how often we should
send RTCP packets at most.

https://bugzilla.gnome.org/show_bug.cgi?id=746543
2015-06-02 11:38:15 +02:00
Sebastian Dröge
565cd49643 rtpsession: Don't crash if we receive FIR/PLI from a source we don't know 2015-05-21 13:26:53 +03:00
Santiago Carot-Nemesio
2fb1fe2ee3 rtpsession: Fix collection of statistics
Stats should be collected on the media rtp source not in the
sender one.

https://bugzilla.gnome.org/show_bug.cgi?id=749669
2015-05-21 12:56:12 +03:00
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
Sebastian Dröge
ca110fb0b8 rtpjitterbuffer: When detecting a huge seqnum gap, wait for 5 consecutive packets before resetting everything
It might just be a late retransmission or spurious packet from elsewhere, but
resetting everything would mean that we will cause a noticeable hickup. Let's
get some confidence first that the sequence numbers changed for whatever
reason.

https://bugzilla.gnome.org/show_bug.cgi?id=747922
2015-05-18 18:43:15 +03:00
Tim-Philipp Müller
2e412a447a docs: update example pipelines in element docs
Mostly gst-launch -> gst-launch-1.0
Use autovideosink/autoaudiosink more often.
Sprinkle some converters here and there.
2015-05-10 11:05:00 +01:00
Sebastian Dröge
27729a2960 Revert "rtpsession: Also report internal sources in on-new-ssrc and on-ssrc-active"
This reverts commit d22ec49632.

Application code might expect that it only gets external sources on those
signals, and get confused by this. If anything we would need to add new
signals.
2015-05-07 14:51:45 +02:00
Sebastian Dröge
d22ec49632 rtpsession: Also report internal sources in on-new-ssrc and on-ssrc-active
Without this it seems impossible for an application to easily get notified
about the internal ssrcs that are created, e.g. sender sources, and also
to know when they are active and produce RTCP packets.

https://bugzilla.gnome.org/show_bug.cgi?id=746747
2015-05-06 11:21:22 +02:00
Sebastian Dröge
9d22ad421b rtpsession: The stats min_interval is in seconds, not nanoseconds
We have to scale it to compare it against our clock times.
2015-05-04 14:12:07 +02:00
Sebastian Dröge
afe1d5a89f rtpsession: Only return TRUE if early feedback was requested already and it's early enough 2015-05-04 14:11:00 +02:00
Sebastian Dröge
73c0c2920f rtpstats: Average RTCP packet size is in bytes, bandwidths in bits
We need to convert the size to bits for our calculations.

https://bugzilla.gnome.org/show_bug.cgi?id=747863
2015-04-27 16:45:40 +02:00
Sebastian Dröge
475b1e607e rtpstats: Use the same lower limit for RTCP bandwidth to stop sending RTCP everywhere
https://bugzilla.gnome.org/show_bug.cgi?id=747863
2015-04-27 16:45:33 +02:00
Sebastian Dröge
7596ed91b8 rtpsession: Use bandwidth calculation by default instead of some arbitrary hardcoded value
https://bugzilla.gnome.org/show_bug.cgi?id=747863
2015-04-27 16:45:25 +02:00
Sebastian Dröge
928cd110bc rtpsession: Bandwidth is supposed to be in bits/s, not bytes/s
https://bugzilla.gnome.org/show_bug.cgi?id=747863
2015-04-27 16:45:14 +02:00
Luis de Bethencourt
9391622579 Rename property enums from ARG_ to PROP_
Property enum items should be named PROP_ for consistency and readability.
2015-04-27 11:22:11 +01:00
Ilya Konstantinov
fd391a5404 rtpjitterbuffer: Fix "stats" property docs
https://bugzilla.gnome.org/show_bug.cgi?id=748436
2015-04-26 21:15:44 +02:00
Tim-Philipp Müller
d753a3eeb1 Remove obsolete Android build cruft
This is not needed any longer.
2015-04-26 17:55:07 +01:00
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
edcc5be297 rtpjitterbuffer: When request retransmissions for future packets, consider the packet spacing in the extra delay
We now take the maximum of 2*jitter and 0.5*packet_spacing for the extra
delay. If jitter is very low, this should prevent unnecessary retransmission
requests to some degree.

https://bugzilla.gnome.org/show_bug.cgi?id=748041
2015-04-22 20:27:18 +02:00
Sebastian Dröge
3fe8ceff14 rtpjitterbuffer: Take a running average of the packet spacings instead of just the latest
https://bugzilla.gnome.org/show_bug.cgi?id=748041
2015-04-22 20:25:43 +02:00
Miguel París Díaz
f81c9a9568 rtpjitterbuffer: Add "rtx-next-seqnum" property
If this is set to FALSE, rtpjitterbuffer will not request retransmissions for
future packets based on when they are estimated to arrive.

See also https://bugzilla.gnome.org/show_bug.cgi?id=748041

https://bugzilla.gnome.org/show_bug.cgi?id=739868
2015-04-22 19:51:18 +02:00
Sebastian Dröge
68dfe93463 rtxreceive: Put debug output for retransmission requests at the right place
Before it was only ever printed once for every time a ssrc was associated with
a specific stream.
2015-04-22 19:51:18 +02:00
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
Arun Raghavan
26bec72e52 rtpsession: Track RTX ssrc caps
This is needed so that we can generate SR for RTX stream correctly (the
clock rate is required).

https://bugzilla.gnome.org/show_bug.cgi?id=747394
2015-04-16 17:33:37 +02:00
Sebastian Dröge
17c6532b75 rtprtxsend: Copy over timestamps from the orignal buffers to the RTX buffers
https://bugzilla.gnome.org/show_bug.cgi?id=747394
2015-04-16 17:33:37 +02:00
Sebastian Dröge
caa255d2ed rtprtx*: Fix typos 2015-04-14 19:08:38 +02:00
Sebastian Dröge
bd19b08d6d rtpsession: Not sending early RTCP now because of dithering means we send it with the next compound packet 2015-04-14 18:42:44 +02:00
Sebastian Dröge
4223d0c114 rtpsession: Improve debug output a bit if we can't allow early feedback 2015-04-14 18:42:44 +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
Miguel París Díaz
c4bb6a098b rtpjitterbuffer: Add "rtx-max-retries" property
This property allows to limit the maximum number of retransmission
for a specific packet.

https://bugzilla.gnome.org/show_bug.cgi?id=739868
2015-04-13 09:09:03 +02:00
Miguel París Díaz
05bd708fc5 rtpjitterbuffer: Fix expected_dts calc in calculate_expected
Right above we consider lost_packet packets, each of them having duration,
as lost and triggered their timers immediately. Below we use expected_dts
to schedule retransmission or schedule lost timers for the packets that
come after expected_dts.

As we just triggered lost_packets packets as lost, there's no point in
scheduling new timers for them and we can just skip over all lost packets.

https://bugzilla.gnome.org/show_bug.cgi?id=739868
2015-04-13 09:06:33 +02:00
Sebastian Dröge
1a2f253c3a rtpjitterbuffer: Make the next output buffer discont after resetting the jitterbuffer
Resetting the jitterbuffer drops all packets and other things, and will cause
a discontinuity in the packets received by the depayloaders. They should now
also flush anything they had pending as the new data will start at a different
position.

https://bugzilla.gnome.org/show_bug.cgi?id=739868
2015-04-13 09:05:34 +02:00
Tim-Philipp Müller
2fde2011b2 docs: make GstRTCPSync enum show up in rtpbin docs
https://bugzilla.gnome.org/show_bug.cgi?id=747358
2015-04-05 20:07:19 +01:00
Nicolas Dufresne
12762ad1a5 rtpjitter: Account for rtx_retry in overflow check
As rtx_retry is part of the substraction, we need to take it into
account, otherwise we may endup with a big value.
2015-03-25 15:25:56 -04:00
Sebastian Dröge
0e3609a6e1 rtpsession: Fix another instance of sticky event misordering warnings
Make sure that the sync_src pad has caps before the segment event.
Otherwise we might get a segment event before caps from the receive
RTCP pad, and then later when receiving RTCP packets will set caps.
This will results in a sticky event misordering warning

This fixes warnings in the rtpaux unit test but also in the
rtpaux and rtx examples in tests/examples/rtp

https://bugzilla.gnome.org/show_bug.cgi?id=746445
2015-03-21 19:30:32 +01:00
Sebastian Dröge
17d90b453f rtpsession: Also start the RTCP send thread when receiving RTP or RTCP
Before we only started it when either:
- there is no send RTP stream
or
- we received an RTP packet for sending

This could mean that if the send RTP pads are connected but never receive any
RTP data, and the same session is also used for receiving RTP/RTCP, we would
never start the RTCP thread and would never send RTCP for the receiving part
of the session.

This can be reproduced with a pipeline like:

gst-launch-1.0 rtpbin name=rtpbin \
udpsrc port=5000 ! "application/x-rtp, media=video, clock-rate=90000, encoding-name=H264" ! rtpbin.recv_rtp_sink_0 \
udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \
rtpbin.send_rtcp_src_0 ! fakesink name=rtcp_fakesink silent=false async=false sync=false \
rtpbin.recv_rtp_src_0_2553225531_96 ! decodebin ! xvimagesink \
fakesrc ! valve drop=true ! rtpbin.send_rtp_sink_0 \
rtpbin.send_rtp_src_0 ! fakesink name=rtp_fakesink silent=false async=false sync=false -v

Before this change the rtcp_fakesink would never send RTCP for the receiving
part of the session (i.e. no receiver reports!), after the change it does.

And before and after this change it would send RTCP for the receiving part of
the session if the sender part was omitted (the last two lines).
2015-03-21 17:38:07 +01:00
Sebastian Dröge
1018aacb35 rtprtxsend: Add support for buffer lists 2015-03-19 11:54:37 +01:00
Sebastian Dröge
57ff27f8c8 rtprtxqueue: Implement support for buffer lists 2015-03-19 11:54:37 +01:00
Ramiro Polla
63944753b0 rtpdtmfmux: properly escape percent sign in documentation 2015-03-14 14:22:26 +00:00
Tim-Philipp Müller
c4fa54da17 Fix double semicolons 2015-03-10 09:31:20 +00:00
Sebastian Dröge
9e934d076b rtpjitterbuffer: Drop packets with sequence numbers before the seqnum-base
These are outside the expected range of sequence numbers and should be
clipped, especially for RTSP they might belong to packets from before a seek
or a previous stream in general.
2015-03-09 11:10:35 +01:00
Sebastian Dröge
38bf3d3808 rtpjitterbuffer: Don't forget to unlock the mutex when receiving GAPs in TCP streams 2015-03-09 10:05:14 +01:00
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
Sebastian Dröge
8984e18ef7 rtpsession: Add explanation why we have space for 32 hash tables
And also create only one, there's no need yet to create all 32 until
we implement RFC2762.
2015-03-04 11:30:43 +01:00
Sebastian Dröge
af2bdd6e15 Revert "rtpsession: Do not use an array of maps if they are not being used"
This reverts commit 1591adf4cd.

https://bugzilla.gnome.org/show_bug.cgi?id=745586#c1:
It's the beginning of an implementation of RFC 2762, which is needed for
large multicast groups. The implementation is not yet complete but why
not leave what is there and implement RFC 2762 instead?
2015-03-04 11:26:57 +01:00
Santiago Carot-Nemesio
1591adf4cd rtpsession: Do not use an array of maps if they are not being used
rtpsession declares an array of maps to store srrcs but only the
the key 0 is being used. This patch replaces the array of maps
for just one map and remove useless parameters in rtpsession

https://bugzilla.gnome.org/show_bug.cgi?id=745586
2015-03-04 11:25:30 +01:00
Sebastian Dröge
939a95d44b rtpsession: Send initial events on sync_rtcp pad when using RTP/RTCP muxing
Otherwise we will just send buffers on the pad without any events beforehand
and will get g_warnings() about that.
2015-02-19 13:34:47 +02:00
Sebastian Dröge
735c6c40f8 rtpjitterbuffer: When resetting the jitterbuffer because of packet discont, don't flush sticky events
We will otherwise flush away STREAM_START, CAPS or SEGMENT events and will
confuse downstream with buffers that come before such events.
2015-02-17 16:57:55 +02:00
Sebastian Dröge
b79eff7f9b rtpsession: Handle first RTCP packet and early feedback correctly
According to RFC 4585 section 3.5.3 step 1 we are not allowed to send
an early RTCP packet for the very first one. It must be a regular one.

Also make sure to not use last_rtcp_send_time in any calculations until
we actually sent an RTCP packet already. In specific this means that we
must not use it for forward reconsideration of the current RTCP send time.
Instead we don't do any forward reconsideration for the first RTCP packet.
2015-02-11 10:32:46 +01:00
Sebastian Dröge
075eb10e65 rtpsession: Fix signal name
This wasn't meant to be pushed at all yet, but now that it's there
already it won't hurt to make it correct at least.
2015-01-30 18:22:31 +01:00
Sebastian Dröge
ec99bbb5e1 rtpstats: Fix typo in documentation 2015-01-30 16:56:35 +01:00
Sebastian Dröge
77511b156e rtpsession: Add new on-receiving-rtcp signal
This will be emitted whenever an RTCP packet is received. Different to
on-feedback-rtcp, this signal gets every complete RTCP packet and not
just the individual feedback packets.
2015-01-30 16:50:36 +01:00
Sebastian Dröge
e4ed852041 rtpsession: Deprecate rtcp-immediate-feedback-threshold property
It had no effect since quite some time and also is not needed in general,
especially not to switch between immediate feedback mode and early feedback
mode. The latest understanding of the RFC is that from the endpoint point of
view, both modes are exactly the same. RTCP is only allowed to use the
bandwidth as given by the RFC constraints, as such it is only ever possible
to schedule a RTCP packet early but it's against the RFC to schedule more RTCP
packets.

The difference between immediate feedback mode and early feedback mode is that
the former guarantees that an RTCP packet can be sent for every event
"immediately", which means that the bandwidth calculations from the RFC have
resulted in an RTCP scheduling interval that is small enough. Early feedback
mode on the other hand means that we can schedule some packets early to make
that happen, but it's not guaranteed at all that it's possible to schedule
an RTCP packet per event (i.e. they need to be accumulated or dropped).
2015-01-26 18:49:31 +01:00
Sebastian Dröge
b07b7736b3 rtpsession: Delay the next regular RTCP packet after early RTCP
This is required to not exceed the short term average RTCP bitrate when
using early feedback as compared to without early feedback.
2015-01-26 18:49:31 +01:00
Sebastian Dröge
bc9111a03d rtpsession: Add new send-rtcp-full signal
This indicates with a boolean return value if scheduling a new RTCP packet
within the requested delay was possible. Otherwise it behaves exactly like
send-rtcp. The only reason for adding a new signal is ABI compatibility.
2015-01-26 18:49:31 +01:00
Sebastian Dröge
60e2d0c84f rtpsession: Fix indention 2015-01-22 11:03:25 +01:00
Sebastian Dröge
87c8c163a8 rtpjitterbuffer: If we get a gap with a buffer without DTS, error out
We (currently?) can't really handle gaps between RTP packets if they're not
properly timestamped. The current code would go into calculations with
GST_CLOCK_TIME_NONE and then cause assertions everywhere. It's probably
better to error out cleanly instead.
2015-01-07 18:05:18 +01:00
Tim-Philipp Müller
c62209d050 rtpptdemux: just drop invalid rtp packets instead of erroring out
Apparently linphone sends an invalid RTP packet as very
first packet. We want to ignore that instead of erroring
out (same for any other invalid packets really).

https://bugzilla.gnome.org/show_bug.cgi?id=741398
2014-12-25 15:48:04 +00:00
Tim-Philipp Müller
bcad30510b rtpptdemux: fix 0.10-ism in docs 2014-12-25 15:44:15 +00:00
Thibault Saunier
52a1773b40 rtpsession: Use an empty iterator in iterate_internal_link when no links
And not a NULL Iterator, so it is consistent with the way it usually
works and avoid user to need a different code paths to handle that.
2014-12-09 20:38:22 +01:00
Thibault Saunier
aa89278ade rtpjitterbuffer: Use an empty iterator in iterate_internal_link when no links
We used to setup an iterator with 1 GValue set with a NULL object
pointer which is not the normal way to do that. Instead we should make
sure that the first call to gst_iterator_next returns GST_ITERATOR_DONE.
2014-12-03 11:17:11 +01:00
Olivier Crête
ccac1f8c0b rtprtxreceive: Use offset when copying header
The header is not always at the start of the packet, so we need to compute
the offset first.
2014-11-29 18:38:12 -05:00
Miguel París Díaz
6daa57868f rtpjitterbuffer: ensure rtx_retry_period >= 0
https://bugzilla.gnome.org/show_bug.cgi?id=739344
2014-11-22 14:48:57 +00:00
Arun Raghavan
45e716e75d rtpbin: Fix up new_jitterbuffer signal prototype 2014-11-20 22:42:59 +05:30
Arun Raghavan
56436ccced rtpbin: Document how to control per-SSRC retransmission 2014-11-20 20:24:42 +05:30
Arun Raghavan
1c3b233fef rtpmanager: Trivial typo fix 2014-11-10 13:16:50 +05:30
Tim-Philipp Müller
d940c21b78 rtpjitterbuffer: implement get/set for new rtx-min-retry-timeout property
Properties are so much more useful if you can actually set
and get their values.
2014-11-02 13:06:33 +00:00
Tim-Philipp Müller
b02d73a0ed rtpjitterbuffer: fix crash on some 32-bit systems
Make sure to pass right number of bits to gst_structure_new()
which is a vararg function.

Fixes elements/rtpaux unit test on ppc32.
2014-10-25 12:45:31 +01:00
Wim Taymans
bd09dc96e9 rtpjitterbuffer: limit the retry frequency
When the RTT and jitter are very low (such as on a local network), the
calculated retransmission timeout is very small. Set some sensible lower
boundary to the timeout by adding a new property. We use the packet
spacing as a lower boundary by default.
2014-10-22 15:04:24 +02:00
Miguel París Díaz
4b5243c43d gstrtpjitterbuffer: add "rtx-min-delay" property
This property is useful to set a min time to wait before sending a
retransmission event.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=735378
2014-10-22 15:00:27 +02:00
Wim Taymans
0b81b316b5 jitterbuffer: Refactor code
Refactor some code dealing with calculating various timeouts.

See https://bugzilla.gnome.org/show_bug.cgi?id=735378
2014-10-22 14:59:57 +02:00
Miguel París Díaz
e6504e3a65 rtpsession: fix Early Feedback Transmission
In early retransmission we are allowed to schedule 1 regular RTCP packet
at an earlier time. When we do that, we need to set allow_early to FALSE
and ignore/drop (or merge) all future requests for early transmission.
We now first check if we can schedule an early RTCP and if we can,
actually prepare the data for the next RTCP interval.

After we send the next regular RTCP after the early RTCP, we set
allow_early to TRUE again to allow more early requests.

Remove the condition for the immediate feedback for now.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738319
2014-10-22 13:13:47 +02:00
Wim Taymans
09f179139d rtpjitterbuffer: make debug line less confusing 2014-10-21 13:10:53 +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
Sjoerd Simons
0ee384b251 rtpmux: Don't set PROXY_CAPS flag on the src pad
rtpmux behaves like a funnel in that it forwards whatever upstream is
sending buffers. So setting proxy caps doesn't make sense as the
upstream don't have to have compatible caps, thus resulting in an empty
caps set as a result of a caps query. Instead set fixed caps just
as funnel does.

https://bugzilla.gnome.org/show_bug.cgi?id=738722
2014-10-21 10:52:00 +02:00
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
Olivier Crête
b3069634bd rtpmux: clock-base and seqnum-base -> timestamp-offset and seqnum-offset
These were renamed in GstRTPBasePayload in 1.0
2014-10-10 18:12:23 -04:00
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
Sanjay NM
26a1344f37 Miscellaneous minor cleanups
Fix redundant variables and assignments,
and unreachable breaks.

https://bugzilla.gnome.org/show_bug.cgi?id=736875
https://bugzilla.gnome.org/show_bug.cgi?id=736876
https://bugzilla.gnome.org/show_bug.cgi?id=736879
https://bugzilla.gnome.org/show_bug.cgi?id=736880
https://bugzilla.gnome.org/show_bug.cgi?id=736881
https://bugzilla.gnome.org/show_bug.cgi?id=736888
https://bugzilla.gnome.org/show_bug.cgi?id=736890
https://bugzilla.gnome.org/show_bug.cgi?id=736892
https://bugzilla.gnome.org/show_bug.cgi?id=736893
https://bugzilla.gnome.org/show_bug.cgi?id=736894
2014-09-24 00:45:31 +01:00
Ognyan Tonchev
f7ae4288a2 rtpbin: do not leak encsink pad in error case
https://bugzilla.gnome.org/show_bug.cgi?id=736807
2014-09-18 12:49:53 +03:00
Youness Alaoui
a98341397d jitterbuffer: Allow rtp caps without clock-rate
The jitterbuffer shouldn't force clock-rate on its sink pad, this will cause a negotiation issue since rtpssrcdemux doesn't have the clock-rate and doesn't add it to the caps. The documentation states that the clock-rate can either be specified through the caps or through the request-pt-map signal, so we must remove clock-rate from the pad templates and we must accept the GST_EVENT_CAPS if the caps don't have the clock-rate.

https://bugzilla.gnome.org/show_bug.cgi?id=734322
2014-08-21 18:32:58 -04:00
Sebastian Rasmussen
1a35bf9647 rtpmux: Unref pad template caps after usage
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=734473
2014-08-08 15:38:32 -03:00
Sebastian Rasmussen
ca22ad8da9 rtpdtmfmux: Avoid taking an unnecessary ref
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=733122
2014-07-16 16:45:31 +02:00
Sebastian Dröge
2f47105129 rtpbin: Don't leak caps 2014-06-29 23:55:19 +02:00
Sebastian Dröge
bbca040336 rtpssrcdemux: Fix compiler warning when compiling with G_DISABLE_ASSERT 2014-06-29 19:59:53 +02:00
Wim Taymans
ca9cfd40dd jitterbuffer: improve SR packet handling
Implement 3 different cases for handling the SR:

 1) we don't have enough timing information to handle the SR packet and
    we need to wait a little for more RTP packets. In that case we keep
    the SR packet around and retry when we get an RTP packet in the
    chain function.

 2) the SR packet has a too old timestamp and should be discarded. It is
    labeled invalid and the last_sr is cleared.

 3) the SR packet is ok and there is enough timing information, proceed
    with processing the SR packet.

Before this patch, case 2) and 1) were handled in the same way,
resulting that SR packets with too old timestamps were checked over and
over again for each RTP packet.
2014-06-25 16:14:46 +02:00
Miguel París Díaz
b22aed9bbc gstrtpssrcdemux: manage ssrc of RTCP RR packets
https://bugzilla.gnome.org/show_bug.cgi?id=731324
2014-06-23 16:23:00 -04:00
Wim Taymans
d004eda79d rtpsession: update last_activity when sending RTP
Also update last_activity when doing something with the internal
source to make sure don't timeout early.

See https://bugzilla.gnome.org/show_bug.cgi?id=730217
2014-05-16 16:55:17 +02:00