George Kiagiadakis
c945200ff2
rtprtxsend: use the GstObject lock instead of a new one
2014-01-15 10:13:11 +01:00
Tim-Philipp Müller
335b619cd5
rtprtxsend: remove duplicate assignment
...
Coverity CID 1151680
2014-01-09 23:55:16 +00:00
Aleix Conchillo Flaqué
441f286e28
rtpbin: remove unused list of decoders
...
remove list of decoders, which are already handled by the list of elements.
https://bugzilla.gnome.org/show_bug.cgi?id=719938
2014-01-08 10:23:52 +01:00
George Kiagiadakis
9226091235
rtprtxreceive: modify to use a payload-type map like rtprtxsend
2014-01-03 20:48:29 +01:00
George Kiagiadakis
c8a04bc7b2
rtprtxsend: do not keep history of packets with an unknown payload type
...
This allows to disable retransmission per payload type by not putting
a certain payload type in the map.
2014-01-03 20:48:29 +01:00
Wim Taymans
130ad1b1fa
rtprtxsend: Allow SSRC-multiplexing and multiple payload types in the original stream
...
Conflicts:
tests/examples/rtp/server-rtpaux.c
2014-01-03 20:48:29 +01:00
George Kiagiadakis
41285697ac
rtprtxsend: Add an rtx-ssrc property to allow external control of the ssrc
...
This is useful when one needs to know the SSRC beforehands, so that it can
be used for SRTP for example.
2014-01-03 20:48:29 +01:00
Wim Taymans
679b5a8682
session: also push EOS event to RTCP srcpad
2014-01-03 20:48:29 +01:00
Wim Taymans
03e4a180da
session: place SSRC in Retransmission event
2014-01-03 20:48:29 +01:00
George Kiagiadakis
0a8b149e9e
rtprtxsend: use a realistic limit for the value of max-size-packets
...
G_MAXINT16 is chosen because if the queue contains more than
G_MAXINT16 packets, seqnum comparison will not work properly.
2014-01-03 20:48:28 +01:00
George Kiagiadakis
51edc07127
rtprtxsend: use a GSequence to implement the buffer queue
...
This has the advantage that searching the queue to find the
buffer with the requested seqnum is done with binary search.
2014-01-03 20:48:28 +01:00
George Kiagiadakis
487fa8c989
rtprtxsend: retransmit packets in the same order as the rtx requests
2014-01-03 20:48:28 +01:00
George Kiagiadakis
7d530ab59f
rtprtxsend: Handle the max_size_time property
...
This property allows you to specify the amount of buffers
to keep in the retransmission queue expressed as time (ms)
instead of buffer count (which is the max_size_buffers property).
2014-01-03 20:48:28 +01:00
George Kiagiadakis
920a55532c
rtprtxsend: keep important buffer information in a private structure
...
This is to avoid mapping a buffer every time we need to read a seqnum
or a timestamp.
2014-01-03 20:48:28 +01:00
Julien Isorce
5a1aa75961
rtpmanager: add new rtprtxsend / rtprtxreceive elements
...
The purpose of the sender RTX object is to keep a history
of RTP packets up to a configurable limit (in time). It will
listen for custom retransmission events from downstream. When
it receives a request for retransmission, it will look up the
requested seqnum in its list of stored packets. If the packet
is available, it will create a RTX packet according to RFC 4588
and send this as an auxiliary stream.
The receiver will listen to the custom retransmission events
from the downstream jitterbuffer and will remember the SSRC1
of the stream and seqnum that was requested. When it sees a
packet with one of the stored seqnum, it associates the SSRC2
of the stream with the SSRC1 of the master stream. From then
on it knows that SSRC2 is the retransmission stream of SSRC1.
This algorithm is stated in RFC 4588. For this algorithm to
work, RFC4588 also states that no two pending retransmission
requests can exist for the same seqnum and different SSRCs or
else it would be impossible to associate the retransmission with
the original requester SSRC.
When the RTX receiver has associated the retransmission packets,
it can depayload and forward them to the source pad of the element.
RTX is SSRC-multiplexed
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711084
2014-01-03 20:47:59 +01:00
Wim Taymans
bb2d37b11d
rtpbin: add some docs about AUX elements
2013-12-31 15:08:49 +01:00
Wim Taymans
d08e05b4ef
rtpbin: add support for AUX sender and receiver
...
AUX elements are elements that can be inserted into the rtpbin
pipeline right before or after 1 or more session elements.
The AUX elements are essential for implementing functionality such
as error correction (FEC) and retransmission (RTX).
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711087
2013-12-31 15:08:48 +01:00
Wim Taymans
ae22c95881
rtpbin: make request_element method internally
...
We can use the same method to create encoder and decoder elements, they
are just internal elements that we create.
2013-12-31 15:08:48 +01:00
Wim Taymans
ee7f41ba2e
rtpsession: internal-ssrc is no longer deprecated
2013-12-30 17:00:45 +01:00
Wim Taymans
e721d26c68
rtpbin: add Since tags
2013-12-30 16:59:20 +01:00
Wim Taymans
5a2bc1405e
rtpbin: add signal for new jitterbuffer
...
Emit a signal when a new jitterbuffer is created so that the app can
have a chance to configure it.
2013-12-30 16:52:28 +01:00
Wim Taymans
3f3b2d0886
rtpbin: handle multiple encoder instances
...
Keep track of elements that are added to multiple sessions and make sure
we only add them to the rtpbin once and that we clean them when no
session refers to them anymore.
2013-12-30 16:28:57 +01:00
Wim Taymans
05c8edc174
rtpbin: fix memory leaks
2013-12-30 15:17:05 +01:00
Wim Taymans
9345c2280a
rtpbin: expect the pads on the encoders
...
Don't use request pads for the encoder elements, the signal handler
should request the pads and make sure they are available with the right
name.
2013-12-30 15:17:05 +01:00
Wim Taymans
cbc80d10dd
rtpbin: request-rtp-encoder are no action signals
...
The request-rtp-encoder signals are not action signals so mark them
correctly and use an accumulator to collect the result value.
2013-12-30 15:17:05 +01:00
George Kiagiadakis
5ddf6a5e32
gstrtpsession: suggest upstream to use the new "internal-ssrc" after a collision
...
When a collision is found on the internal ssrc, we have to change it.
Ideally, we want also the payloader upstream to follow this change and use
the new internal ssrc. Ideally we want this condition to be always met:
if there is one payloader sending on this session, its ssrc should match the
internal ssrc.
2013-12-30 14:03:05 +01:00
George Kiagiadakis
17517ca491
rtpsession: allow setting internal-ssrc again
2013-12-30 14:03:05 +01:00
Aleix Conchillo Flaqué
47c65fc269
rtpbin: allow dynamic RTP/RTCP encoders/decoders
...
* gst/rtpmanager/gstrtpbin.[ch]: four new action signals have been
added (request-rtp-encoder, request-rtp-decoder, request-rtcp-encoder
and request-rtcp-decoder). The user will be able to provide encoders
or decoders dynamically. The encoders must follow the srtpenc API and
the decoders the srtpdec API. Having separate signals for RTP and RTCP
allows the user to use different encoders/decoders or provide the same
one (e.g. that would be the case for srtpenc).
Also, rtpbin now allows application/x-srtp in its pads.
https://bugzilla.gnome.org/show_bug.cgi?id=719938
2013-12-30 11:24:00 +01:00
Wim Taymans
f48bbabafc
rtpjitterbuffer: dynamically recalculate RTX parameters
...
Use the round-trip-time and average jitter to dynamically calculate the
retransmission interval and expected packet arrival time.
Based on patches from Torrie Fischer <torrie.fischer@collabora.co.uk>
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711412
2013-12-30 11:18:51 +01:00
Wim Taymans
416bd9a2c3
rtpjitterbuffer: calculate average jitter
2013-12-30 11:18:51 +01:00
Wim Taymans
7181a21ca9
rtpsession: use RTT from the Retransmission event
...
Place the estimated RTT in the Retransmission event and let the session
manager use that instead of the hardcoded value.
2013-12-30 11:18:50 +01:00
Wim Taymans
e996f73d0c
jitterbuffer: take more accurate running-time for NACK
...
Don't use the current time calculated from the tmieout loop for when we
last scheduled the NACK because it might be unscheduled because of a max
packet misorder and then we don't accurately calculate the current time.
Instead, take the current element running time using the clock.
2013-12-30 11:18:50 +01:00
Olivier Crête
ada6ea668b
rtpsession: Add error message if the app tries to set the internal-ssrc
2013-12-13 17:36:36 -05:00
Olivier Crête
d715010d78
rtpsession: Only count nacks when a nack packet is received
...
Not when any RTCP feedback packet is.
2013-12-13 16:08:35 -05:00
Olivier Crête
7af9fdbca6
rtpsession: Process PSFB FIR requests which lack the media ssrc
...
According to RFC 5104 section 4.3.1.2, RTCP PSFB FIR message SHALL
have a media_ssrc field set to 0. The actual media ssrc is in the FCI.
So in that case, we ignore the retained feedback and just let it through
to the rtp_session_process_fir() function which will check for the actual
SSRC inside the FCI.
Fixes a regression introduced by commit 57c27ec3
2013-12-13 16:01:07 -05:00
George Kiagiadakis
6a2de911fa
rtpsession: fix rb blocks disappearing after the first rtcp cycle with multiple senders
...
Previously, when the session had multiple internal sender SSRCs, it would
issue SR reports with RB blocks only on the first RTCP timeout and afterwards
SR reports would be sent empty. This was because the "generation" number
in RTPSource would increase more than once during the same cycle and afterwards
it would always be greater than the session's generation, which would cause
it to be skipped from being included in RBs.
This commit fixes this problem by:
1) Increasing the RTPSource generation only at the end of each cycle,
which essentially fixes the problem but only when the internal senders
are less than GST_RTCP_MAX_RB_COUNT.
2) Keeping for each RTPSource a set of SSRCs which stores which SSRC's
SR the given RTPSource has been reported in, which also fixes the problem
when the internal senders are more than GST_RTCP_MAX_RB_COUNT. This is
necessary because of the fact that any RTPSource is marked as reported
in itself's SR and makes it impossible to know if it has been reported
in other SRs too or not, and which.
2013-12-12 16:44:27 +01:00
George Kiagiadakis
c78a115154
rtpsession: keep extra stats for scheduling BYE
...
Keep an extra stats structure for scheduling the BYE packets. When we
decide to schedule BYE, make a copy of the current stats into the
bye_stats. Then while we schedule the BYE, update and use only the
bye_stats. When we finished scheduling the BYE packet, we use the
regular stats again.
2013-12-12 10:38:43 +01:00
George Kiagiadakis
282028e753
rtpsession: when we schedule BYE, only deal with BYE sources
...
When we are doing the RTCP timeout to schedule BYE packets, don't
generate RTCP for all sources but only for the sources marked as BYE.
2013-12-12 10:34:38 +01:00
George Kiagiadakis
6a421c3d81
rtpsession: reset state after scheduling BYE
...
After we do RTCP, we are not scheduling bye anymore.
2013-12-12 10:32:48 +01:00
George Kiagiadakis
0a0ff100ef
rtpsession: also count NACKS when no signal was pending
2013-12-12 10:31:38 +01:00
George Kiagiadakis
bec9c04ea0
session: ignore RTCP packets for the BYE sources
...
When we are scheduling BYE packets, ignore all RTCP for the sources that
are scheduling a BYE packet. Other sources that are not scheduling BYE
should continue receiving RTCP packets as usual.
2013-12-12 10:09:25 +01:00
Julien Isorce
33b398e345
rtpsession: determine if the session is doing point-to-point
...
In this case T_dither_max is set to 0 according to RFC 4585
2013-12-10 16:57:56 +01:00
Wim Taymans
eee515cb2c
rtpjitterbuffer: serialize events in the buffer
...
Serialize events into the jitterbuffer by inserting them with a -1
seqnum.
Update unit test to expect events from the streaming thread.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=652986
2013-12-10 11:57:37 +01:00
Wim Taymans
36e78bc5ca
rtpjitterbuffer: detect -1 seqnum
...
Keep the seqnum as a full guint so that we can check for -1 entries and
deal with them correctly.
Immediately try to push -1 seqnum.
2013-12-10 11:04:06 +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
e8edecc56e
rtpsession: don't unref buffer twice
...
Cleaning the packet info will already unref the buffer.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715078
2013-11-28 16:51:13 +01:00
Wim Taymans
710d1f3a2a
rtpjitterbuffer: improve clear-pt-map handling
...
Don't reset the expected output seqnum when clearing the pt map because this
could stall the jitterbuffer forever.
Add a unit test for this.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=709800
2013-11-25 15:52:22 +01:00
Tim-Philipp Müller
901ec63462
rtpjitterbuffer: fix wake-up when new buffers come in after running empty
...
Spotted by 'gratias' on IRC. Probably introduced in recent refactoring.
https://bugzilla.gnome.org/show_bug.cgi?id=715039
2013-11-25 00:37:50 +00:00
Wim Taymans
4c9474905b
rtpjitterbuffer: pass downstream flowreturn to upstream
...
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712722
2013-11-22 12:27:31 +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
Tim-Philipp Müller
d506409af5
docs: get rid of 'Since: 0.10.x' markers
...
And some gtk-doc markup fixes.
2013-11-18 14:47:35 +00:00
Tim-Philipp Müller
548e756e0a
rtpmanager: fix Since markers
...
Should be next stable release series version
2013-11-16 12:15:14 +00:00
George Kiagiadakis
387e3b918a
rtpjitterbuffer: Fix stats property field names and documentation
2013-11-15 16:23:34 +02:00
Torrie Fischer
acf74435e3
gstrtpsession: Implement a number of feedback packet statistics
...
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711693
2013-11-15 15:21:19 +01:00
Wim Taymans
b450d31503
rtpjitterbuffer: rename property to 'stats'
...
This makes the unit test work.
We can later also add more stats, not specific to retransmission.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711411
2013-11-14 09:24:26 +01:00
Torrie Fischer
22ceb80ba9
rtpjitterbuffer: implement rtx statistics
2013-11-14 09:24:26 +01:00
Wim Taymans
2e5b462ae3
jitterbuffer: advance expected seqnum after dropping
...
After dropping a buffer, move our expected seqnum
Conflicts:
gst/rtpmanager/gstrtpjitterbuffer.c
2013-11-13 12:02:57 +01:00
Wim Taymans
e4bc81d7d2
rtpsession: remove collision reconfigure event
...
Remove bogus reconfigure event on collision, we don't want to send the event on
the receiving RTP pad and the collision event is now handling this
case.
See https://bugzilla.gnome.org/show_bug.cgi?id=711560
2013-11-11 15:27:18 +01:00
Julien Isorce
b32fc6f416
gstrtpsession: send custom upstream event "GstRTPCollision" on send_rtp_sink pad
...
See https://bugzilla.gnome.org/show_bug.cgi?id=711560
2013-11-11 15:25:52 +01:00
George Kiagiadakis
b81b2efa3e
rtpjitterbuffer: fix crash when do-retransmission=true and a lot of buffers are lost
...
The problem here was that the jitterbuffer lock was unlocked to push
the event, but that caused another thread to remove the timer currently
being processed, probably because the amount of rtx events
(and therefore timers) was getting too high. The solution is to
unlock and push the event only after timer processing has finished.
fixes https://bugzilla.gnome.org/show_bug.cgi?id=711131
2013-11-11 11:51:45 +01:00
Wim Taymans
c8db05d610
rtpsource: update receiver stats for sender
...
An internal sender in a session is also a receiver of its own packets so update
the receiver stats. Other senders in the session will use this info to generate
correct RB blocks in their SR reports.
2013-11-07 16:24:30 +01:00
Wim Taymans
268a75e705
rtpsource: refactor receiver stats update
2013-11-07 16:24:30 +01:00
Wim Taymans
e96f8f519c
rtspsrc: proxy new buffer mode
2013-10-31 10:38:35 +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
d4892859d4
jitterbuffer: fix race in flush-start/flush-stop
...
When flush-stop arrives before we process the result of the _push() in the
loop function, we might pause even though we are not flushing anymore. Fix this
race by waiting for the srcpad loop function to completely pause after doing the
flush-start.
2013-10-04 12:35:18 +02:00
Wim Taymans
00056965e8
rtpjitterbuffer: fix race when updating the next_seqnum
...
If we were not waiting for the missing seqnum when we insert the lost packet
event in the jitterbuffer, we end up not updating the next_seqnum and wait
forever for the lost packets to arrive. Instead, keep track of the amount of
packets contained by the jitterbuffer item and update the next expected
seqnum only after pushing the buffer/event. This makes sure we correctly handle
GAPS in the sequence numbers.
2013-09-30 12:31:00 +02:00
Wim Taymans
fde438791e
rtpjitterbuffer: small debug improvement
2013-09-30 12:30:23 +02:00
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
03d520eb69
rtpjitterbuffer: pause timer when PAUSED
...
Also pause the timer when we go to the PAUSED state. It is possible that we
don't have a clock or base-time in PAUSED to perform the timeouts.
2013-09-30 11:16:32 +02:00
Wim Taymans
4a31aec513
rtpjitterbuffer: improve debug
2013-09-30 11:15:25 +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
2efd58fc84
rtpbin: avoid some pad link checks
...
Link pads without checks, we know it will work.
2013-09-25 17:38:31 +02:00
Wim Taymans
97f4674655
rtpjitterbuffer: calculate some stats
2013-09-25 10:50:05 +02:00
Wim Taymans
b1d29483bb
rtpjitterbuffer: move send_lost_event function
...
Move the send_lost_event function to the do_lost_event handling, there is no
need to have a separate function.
2013-09-25 10:50:05 +02:00
Wim Taymans
adf5d96044
rtpmanager: update docs
2013-09-23 16:34:15 +02:00
Wim Taymans
e5019de80d
docs: update docs with 1.0 element names
2013-09-23 15:36:47 +02:00
Wim Taymans
8ce674da87
rtpjitterbuffer: always store lost event in jitterbuffer
...
Always prepare a lost event in the jitterbuffer, it is to wake up and make the
pushing thread continue. We drop the event when we are not supposed to push lost
events downstream.
2013-09-23 14:45:27 +02:00
Wim Taymans
9f3345fcc2
rtpjitterbuffer: schedule lost event differently
...
Schedule the lost event by placing it inside the jitterbuffer with the seqnum
that was lost so that the pushing thread can interleave and push it properly.
2013-09-23 14:45:27 +02:00
Wim Taymans
ae389aeb0c
rtpjitterbuffer: remove list debug
2013-09-23 14:45:26 +02:00
Wim Taymans
28641e3145
rtpjitterbuffer: add type to the item
...
So that the upper layer can know what data is contained in the item.
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
1760817005
rtpjitterbuffer: update expected timer when possible
...
When we receive a packet and we have some missing packets, we can update their
estimated arrival times based on the timestamp difference.
2013-09-23 14:45:25 +02:00
Wim Taymans
fdc1ed1680
rtpjitterbuffer: fix order of timeout events
...
Improve the order of the timeout events, if there are timers with the same
timeout, we want to trigger the lowest seqnum first. For this we need to loop
over the complete array of timers to find the best one before triggering the
timeout.
2013-09-23 14:45:25 +02:00
Wim Taymans
0b1a7edfea
rtpjitterbuffer: send lost event before signaling next buffer
...
First send the lost event, then update the next_seqnum counter and then
send the signal to the pushing thread that it can retry to push a buffer. This
avoids pushing out buffers before the lost event is pushed.
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
3c421e7e48
rtpjitterbuffer: add option to reset retransmission timers
2013-09-23 14:45:24 +02:00
Wim Taymans
6f4deab298
rtpjitterbuffer: stop the timer thread
...
The timeout code could release the lock so we need to check if we are allowed to
wait for the clock some more.
2013-09-23 14:45:24 +02:00
Wim Taymans
cba4e6a707
rtpjitterbuffer: unlock only once
2013-09-23 14:45:24 +02:00
Wim Taymans
5dc207948c
rtpjitterbuffer: improve flush and shutdown
...
There is no need to unschedule the timer in flush-start, flush-stop will remove
the timers and unschedule.
Unschedule the current timer before attempting to join the timer thread.
2013-09-23 14:45:23 +02:00
Wim Taymans
a512cc2d3c
rtpjitterbuffer: set correct expected time
...
When we already have a timer for a packet, skip it but don't forget to adjust
the dts to the expected dts of the next packet.
2013-09-23 14:45:23 +02:00
Wim Taymans
517ea0f4d9
jitterbuffer: improve debug
2013-09-23 14:45:23 +02:00
Wim Taymans
fd6c57cf10
rtpjitterbuffer: keep delay as a separate variable in timer
...
Keep a separate delay in the timer so that we still know the original timestamp
of the packet that this timer refers to. We can then place the correct
running-time in the Retransmission event.
2013-09-19 14:32:48 +02:00
Wim Taymans
d34184dd03
rtpjitterbuffer: fix writability of properties
2013-09-19 14:32:48 +02:00
Wim Taymans
6bb2626498
rtpjitterbuffer: reevaluate the current timer after timeout
...
When we trigger the timeout logic of a timer, reevaluate it because it is
possible that it still has the lowest timeout.
2013-09-18 16:33:40 +02:00
Wim Taymans
8d021b6ede
rtpjitterbuffer: don't update time when unscheduled
...
Don't try to estimate the current time when we got unscheduled.
2013-09-18 16:31:26 +02:00
Wim Taymans
65606a25bf
rtpjitterbuffer: init packet spacing on first buffer
...
Already init the packet spacing variables on the first buffer so that we can
calculate the spacing on the second buffer already.
2013-09-18 16:29:37 +02:00
Wim Taymans
f2efdf28f5
rtpjitterbuffer: push the lost event from the timer thread
...
Instead of pushing the lost event from the chain function, schedule a timeout
that will push the lost event from the timer thread. This avoid blocking the
upstream thread while we push and sync the event.
2013-09-18 14:57:09 +02:00
Wim Taymans
5d5fc03e04
rtpjitterbuffer: round gap duration to multiple of duration
...
Make sure the gap duration in the lost event is a multiple of the packet
duration.
Enable another test.
2013-09-18 14:12:47 +02:00
Wim Taymans
6e4a051d40
rtpjitterbuffer: keep track of duration
...
Keep track of the estimated duration of missing packets and use it in the lost
event.
Enable another unit test
2013-09-18 12:29:38 +02:00
Wim Taymans
ac3bb3acf6
rtpjitterbuffer: handle large gaps with one lost event
...
When we have a large number of missing packets, generate one lost event for all
the packets that have no chance of being pushed out in time.
Fix and activate unit test for large gaps.
2013-09-18 11:59:28 +02:00
Wim Taymans
26402e1c21
rtpjitterbuffer: refactor lost event sending
...
Also make sure we only increment the expected seqnum and last
output timestamp.
2013-09-18 11:57:06 +02:00
Wim Taymans
f49981a597
jitterbuffer: refactor timeout triggers
2013-09-17 23:29:56 +02:00
Wim Taymans
047021c443
jitterbuffer: simplify the timeout code
...
Keep track of the current time in the timeout loop.
Loop over all timers and trigger all the expired ones, we can do this in the
same loop that selects the new best timer.
2013-09-17 23:29:56 +02:00
Wim Taymans
fa1ef3328b
jitterbuffer: rearrange timer update code
...
Also update the timers when retransmission is disabled. We need to
do this because when we added LOST timers when we detected missing packets and
we need to remove those timers when the packet finally arrives.
2013-09-17 23:29:56 +02:00
Wim Taymans
232fdd8b56
jitterbuffer: release lock on shutdown
2013-09-17 15:19:42 +02:00
Wim Taymans
4de919a17a
jitterbuffer: use separate thread for timeouts
...
Use a separate thread for scheduling the timeouts instead of using the
downstream streaming thread that might block at any time.
2013-09-16 15:55:55 +02:00
Olivier Crête
b9ceafe5af
rtpsession: Demux RTCP buffers from the RTP stream
...
If there are RTCP buffers in the RTP stream, process them as
RTCP. This way, we want receive streams following RFC 5761
https://bugzilla.gnome.org/show_bug.cgi?id=687657
2013-09-13 16:25:49 +02:00
Wim Taymans
28e5f90988
rtpbin: use PacketInfo for the sender
...
Avoid mapping the packet multiple times when sending RTP.
2013-09-13 14:34:28 +02:00
Wim Taymans
a02c9473d8
rtpbin: store more in the PacketInfo
...
Store all info in the PacketInfo so that we can avoid mapping the packet
multiple times.
2013-09-13 14:34:28 +02:00
Wim Taymans
e5c789abd6
session: store more in the PacketInfo structure
2013-09-13 14:34:28 +02:00
Wim Taymans
47662f9ca4
rtpbin: RTPArrivalStats -> RTPPacketInfo
...
Rename a structure because we are also going to use this for the sender
bits.
2013-09-13 14:34:28 +02:00
Wim Taymans
c795b72988
source: small cleanups
2013-09-13 14:34:27 +02:00
Wim Taymans
9f9ba21404
jitterbuffer: handle segments with non-0 start
...
We keep the DTS and PTS in running-time inside the jitterbuffer. Make sure to
transform it back to a buffer timestamp before pushing out the buffer.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=707931
2013-09-12 15:04:30 +02:00
Wim Taymans
f1106cde66
session: only update next check time when reconsidering
...
Don't update the next RTCP check time in all cases but only when we
reconsidered. This avoids delaying sending a full RTCP packet when we
are doing early feedback.
2013-08-27 09:55:52 +02:00
Wim Taymans
47065db0b6
session: add more debug
2013-08-27 09:55:52 +02:00
Wim Taymans
454d75951e
jitterbuffer: fix types of the retransmission event
2013-08-27 09:55:52 +02:00
Wim Taymans
dd4af0d11c
jitterbuffer: only timeout EXPECTED timers on gap
...
Only timeout the EXPECTED timers when we detect a large seqnum gap.
2013-08-27 09:44:18 +02:00
Wim Taymans
4b7bcc2ec1
rtsession: fix locking
...
We need to take the session lock when getting and manipulating the
source.
2013-08-26 11:50:27 +02:00
Wim Taymans
3f46527f75
rtpsession: add some more debug
2013-08-26 11:50:13 +02:00
Wim Taymans
54e7e7547a
rtxqueue: add property to configure queue size
2013-08-23 15:47:25 +02:00
Wim Taymans
84833bed11
rtpbin: proxy jitterbuffer do-retransmission property
2013-08-23 12:10:19 +02:00
Wim Taymans
89b9019e3e
rtx: various improvements
...
Use locking
Don't push from the event handler, collected packets in a queue and push from
the chain function.
Clear queues on shutdown.
2013-08-21 17:02:27 +02:00
Wim Taymans
ee15bc9284
session: generate events correctly
...
Do correct shifting of the bitmask for lost packets.
2013-08-21 17:02:27 +02:00
Wim Taymans
67523d3ecb
rtp: register rtx element better
2013-08-21 17:02:26 +02:00
Wim Taymans
587dc055e9
jitterbuffer: handle EOS
...
When the queue is empty, and we received EOS, pause and push an EOS
event downstream.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706387
2013-08-20 14:36:59 +02:00
Wim Taymans
533f26fc99
jitterbuffer: update docs
2013-08-20 10:26:15 +02:00
Wim Taymans
c7f9ef8012
jitterbuffer: update all timers
...
Keep looping over all registered timers so that we can mark them lost instead of
stopping as soon as we find the timer for the current seqnum.
2013-08-20 10:25:17 +02:00
Wim Taymans
5debda9ca1
jitterbuffer: remove unused variables
2013-08-20 08:55:50 +02:00
Wim Taymans
a88db5fa2c
jitterbuffer: reorganize timer handling
...
Restructure handling of incomming packet and the gap with the expected seqnum
and register all timers from the _chain function.
Convert a timer to a LOST packet timer when the max amount of retransmission
requests has been reached.
2013-08-19 22:04:51 +02:00
Wim Taymans
d9d6eac4bb
jitterbuffer: refactor packet spacing calculation
2013-08-19 22:04:50 +02:00
Wim Taymans
c4dc159656
jitterbuffer: keep track of last seqnum and dts
2013-08-19 22:04:50 +02:00
Wim Taymans
652ce95ca6
jitterbuffer: small cleanups
2013-08-19 22:04:50 +02:00
Wim Taymans
b4a35bbe82
jitterbuffer: reset retransmission timers in add/reschedule
...
Reset the retransmission timers when adding and rescheduling a timer.
2013-08-19 22:04:50 +02:00
Wim Taymans
cf8a0652f3
jitterbuffer: rename variables for packet spacing
2013-08-19 22:04:50 +02:00
Wim Taymans
ec82e4ab7c
jitterbuffer: remove lost timer when we get the packet
...
When we receive a packet, also remove the LOST timer for it.
2013-08-19 22:04:50 +02:00
Wim Taymans
2f03b43b21
jitterbuffer: expected seqnum must increase
...
Only update the expected seqnum when it is bigger than the previous expected
seqnum.
2013-08-19 22:04:50 +02:00
Wim Taymans
c5bf376bb5
jitterbuffer: add more debug
2013-08-19 22:04:50 +02:00
Wim Taymans
ff825a2919
rtxqueue: add retransmission queue element
2013-08-19 22:04:50 +02:00
Wim Taymans
5fe18ee432
session: add some docs
2013-08-19 22:04:49 +02:00
Wim Taymans
482dacfb54
session: handle NACK feedback and generate events
...
Handle and parse the feedback NACK packets and generate a Retransmission
event for each NACKed packet
2013-08-19 22:04:49 +02:00
Wim Taymans
f11c2c9b3b
jitterbuffer: forward flush before stopping dataflow
...
First forward the flush event and then stop our loop function.
2013-08-14 16:19:32 +02:00
Wim Taymans
48174164eb
session: add NACK feedback in RTCP
2013-08-06 15:50:19 +02:00
Wim Taymans
4379ed1dee
source: add methods to register NACK
...
Add a method to register a missing packet for an ssrc along with
methods to get the missing packets and clear them.
2013-08-06 15:50:19 +02:00
Wim Taymans
50638b8106
session: handle Retransmission event and schedule NACK
...
Handle the retransmission event from downstream and use it to schedule a NACK
request.
2013-08-06 15:50:19 +02:00
Wim Taymans
0bddbd682d
session: pass data to remove func
...
Pass the data to the remove function because we are going to deref it when there
is pli or fir.
2013-08-06 15:50:19 +02:00
Wim Taymans
3c82de59f9
session: use common send_rtcp method
...
Reuse the send_rtcp method that already asks for the current time when
requesting a keyframe.
2013-08-05 15:02:59 +02:00
Wim Taymans
3c14c6021c
session: Don't use ClockTimeDiff for unsigned delays
2013-08-05 15:02:59 +02:00
Tim-Philipp Müller
7469cd3a4c
rtpmanager: use generic marshaller
2013-08-04 11:03:07 +01:00
Wim Taymans
7584f91f31
jitterbuffer: send event in right direction
2013-08-04 00:24:36 +02:00
Wim Taymans
9613e481ad
session: add FIR and PLI like other RTCP packets
...
Add the FIR and PLI packets like the other RTCP packet instead of from the
on-sending-rtcp default signal handler.
2013-08-03 00:33:24 +02:00
Wim Taymans
743e1b1191
jitterbuffer: fix property ranges
2013-08-02 17:22:55 +02:00
Wim Taymans
cd0164f4cc
jitterbuffer: push retransmission events
2013-08-02 16:43:59 +02:00
Wim Taymans
9a13267e85
jitterbuffer: add support for retransmission retry
...
When we didn't receive a packet after requesting retransmission, retry
asking for retransmission for a certain period.
2013-08-02 14:54:56 +02:00
Wim Taymans
e9ad5126db
jitterbuffer: add properties
...
Add properties to control retransmission parameters
2013-08-02 14:47:56 +02:00
Wim Taymans
a8c7ff7489
jitterbuffer: use corrected timeout when rescheduling
...
When we recalculate the timeout, use the corrected timeout value depending on
the timer type.
2013-08-02 12:44:58 +02:00
Wim Taymans
9c7e3e3455
jitterbuffer: update timers after queueing
...
Else we might update the timer needlessly for duplicates.
2013-08-02 12:43:00 +02:00
Wim Taymans
ebd6b8f8ab
jitterbuffer: move method up
2013-08-02 12:42:08 +02:00
Wim Taymans
f6b6797874
jitterbuffer: small cleanup
2013-08-02 06:28:32 +02:00
Wim Taymans
0e41414926
jitterbuffer: unschedule old expected packets
...
When we receive a new packet, unschedule old outstanding packets when their
seqnum is too far away.
2013-08-01 23:36:07 +02:00
Wim Taymans
70695466ed
jitterbuffer: refactor timer update
2013-08-01 23:32:00 +02:00
Wim Taymans
4ab3f5d3da
jitterbuffer: update timers when removing
...
Update the timers when we remove a timer.
Handle canceled timers, make them unschedule the current timer and
trigger the timeout code.
2013-08-01 23:24:29 +02:00
Wim Taymans
b983cf675b
jitterbuffer: fix typo
2013-08-01 23:22:02 +02:00
Wim Taymans
f3c658cbe6
jitterbuffer: improve timeout management
...
If we change the seqnum of an existing timer and we were waiting for
that timer, unschedule it. If we change the timeout of an existing timer and we
were waiting on it, only unschedule when the new time is smaller.
2013-08-01 15:40:52 +02:00
Wim Taymans
77e5d320ab
jitterbuffer: install timer for expected arrival
...
Install a timer that is triggered when the expected arrival time of a packet
expired.
2013-08-01 15:11:13 +02:00
Wim Taymans
f08d98404e
jitterbuffer: improve unschedule of timers
...
Conflicts:
gst/rtpmanager/gstrtpjitterbuffer.c
2013-08-01 14:57:11 +02:00
Wim Taymans
9d3b824e2a
jitterbuffer: move code around
2013-08-01 12:21:53 +02:00
Wim Taymans
fe32e80c92
jitterbuffer: estimate inter packet spacing
...
When we see two packets with consecutive seqnums and a different RTP time, use
the DTS difference as the inter packet spacing estimate.
2013-08-01 12:07:11 +02:00
Wim Taymans
255b7106f5
jitterbuffer: keep track of current timeout
2013-08-01 12:01:15 +02:00
Wim Taymans
7e43dba19b
jitterbuffer: cleanup timer handling
2013-08-01 11:49:10 +02:00
Wim Taymans
9d88ac9cbb
jitterbuffer: reset is only possible with a GAP
2013-08-01 11:40:41 +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
80c5934290
jitterbuffer: rename timout variable
2013-08-01 11:14:12 +02:00
Wim Taymans
aa951433ee
jitterbuffer: small cleanup
2013-07-31 17:08:58 +02:00
Wim Taymans
69c78f72d5
jitterbuffer: block output in paused or buffering
2013-07-31 16:59:58 +02:00
Wim Taymans
4fbbc53a49
jitterbuffer: store pts in timer
...
Only store the pts in the timer so that we can both do timeouts with timings on
the input and output of the jitterbuffer.
2013-07-31 16:59:09 +02:00
Wim Taymans
77846d35c6
rtpjitterbuffer: refactor jitterbuffer
...
Refactor the jitterbuffer code. Make separate function for peeking a buffer,
pushing the next buffer, waiting for timeouts and handling the timeouts.
The main loop now tries to push as many buffers as it can until it runs out of
buffers or when it detects a seqnum discont. Then it will wait for some event to
happen before attempting to push more buffers.
Make methods to register timeouts in an array. These timeouts are registered
when we detect a missing packet, sync for the first packet or when we find an
estimation for the end-of-stream.
This greatly simplifies and clarifies the code and also makes it possible to
register more complicated timeout schemes later.
2013-07-30 23:24:23 +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
Wim Taymans
b3e8a85a54
jitterbuffer: refactor
...
Move eos estimation into separate function
2013-07-30 23:24:22 +02:00
Wim Taymans
02359f9219
session: don't make buffer writable prematurely
...
There is no reason to make the SR buffer writable at this point. This is better
delayed until needed.
2013-07-26 22:31:41 +02:00
Wim Taymans
0261199fc4
session: ignore RTCP for inactive sources
2013-07-26 22:31:23 +02:00
Wim Taymans
a4b4ca53c0
session: small cleanup
2013-07-26 22:25:17 +02:00
Wim Taymans
e0abd2e9b5
session: handle partial RTCP report blocks
...
When we have more SSRCs to report than what fit in an RTCP packet, use a
generation counter to make sure all of them end up in a packet eventually.
2013-07-26 17:29:10 +02:00
Wim Taymans
6cce6fb04c
session: create SSRC before doing session cleanup
...
Make the internal source before we do session cleanup
2013-07-26 17:29:10 +02:00
Wim Taymans
5b0298c63e
session: reorganize the report block code
2013-07-26 17:29:10 +02:00
Wim Taymans
3c44cd7c83
session: refactor active and sender checks
2013-07-26 14:21:40 +02:00
Wim Taymans
e952f7ba43
session: remove internal sources on timeout
...
When an internal source times out and becomes a receiver, remove it.
2013-07-26 12:18:01 +02:00
Wim Taymans
e9e2fe3950
session: create an internal source for RTCP
...
When we need to do RTCP and we don't have an internal source yet,
make one.
2013-07-26 12:18:01 +02:00
Wim Taymans
bd0709c15c
session: remove old code to change SSRC
...
Remove code used to change the SSRC after a collision. We now send
a RECONFIGURE event upstream to make the upstream element change the SSRC.
2013-07-26 12:18:01 +02:00
Wim Taymans
88f5a5f355
source: don't update packet SSRC
...
Remove the code to update the SSRC in packets, it can never be called now that
we always use a source with matching packet SSRC.
2013-07-26 12:18:01 +02:00
Wim Taymans
abc90da1dc
session: delay allocation of internal source
...
Allocate the internal source when we receive a caps with the SSRC or when we see
a buffer with the SSRC.
2013-07-26 12:18:01 +02:00
Wim Taymans
e0a1ce1291
session: generate reconfigure on collision
...
When we detect a collision, change the SSRC that we suggest upstream
and trigger RECONFIGURE. This should make upstream select a new SSRC.
2013-07-26 12:18:01 +02:00
Wim Taymans
495d43c089
session: produce RTCP for all internal sources
...
Loop over all the internal sources and produce RTCP. We also need
to queue the RTCP packets and send them when we are finished.
2013-07-26 12:18:00 +02:00
Wim Taymans
9505fd4150
session: deprecate internal source and ssrc properties
...
Deprecate the internal source and internal ssrc properties. There might
be more than one internal source.
2013-07-26 12:17:59 +02:00
Wim Taymans
3d6ee1fb5e
session: internal sources don't use probation
2013-07-26 12:17:59 +02:00
Wim Taymans
0e53e9109e
session: give caps to session
...
Let the session parse the caps and update its SSRC when needed.
2013-07-26 12:17:59 +02:00
Wim Taymans
c06482a2cb
session: make method to suggest available SSRC
...
Make a method to suggest the best available SSRC. This is the SSRC of the last
created internal source and is used to instruct upstream to produce this
SSRC.
2013-07-26 12:17:59 +02:00
Wim Taymans
33ce50e8b1
session: keep SDES and set on new internal sources
...
Keep track of the SDES ourselves and set it on all newly created
internal sources.
2013-07-26 12:17:59 +02:00
Wim Taymans
5652f02b76
session: make method to make internal sources
...
Add a method to obtain an internal source and use it to create
our internal source
2013-07-26 12:17:59 +02:00
Wim Taymans
7f83927c95
session: count internal sources and how many are senders
2013-07-26 12:17:58 +02:00
Wim Taymans
719343c206
rtpsession: separate BYE marking and scheduling
...
First mark sources with BYE and then schedule the BYE RTCP message.
2013-07-26 12:17:58 +02:00
Wim Taymans
391943ba82
session: get SSRC from RTCP packet itself
...
Get the SSRC from the RTCP packet instead.
2013-07-26 12:17:57 +02:00
Wim Taymans
a3f75a17ef
session: fix bandwidth calculation
...
We iterate over all sources and the internal one is also in the
hashtable so avoid adding it twice.
2013-07-26 12:17:57 +02:00
Wim Taymans
9eaef9d332
session: add some docs
2013-07-26 12:17:56 +02:00
Wim Taymans
2163355a47
session: Rearrange RTCP reporting a little
...
Make a function to generate an RTCP packet for a source, pass the source as a
parameter.
Move timeout of collisions to session cleanup phase.
2013-07-26 12:17:56 +02:00
Wim Taymans
a3bf374351
session: move check for is_early around
...
Move the check for the early RTCP to where it is needed and used.
2013-07-26 12:17:56 +02:00
Wim Taymans
b069db6a2e
session: parse packet outside of the session lock
2013-07-26 12:17:56 +02:00
Wim Taymans
57c27ec319
session: do nicer checks for internal sources
2013-07-26 12:17:56 +02:00
Wim Taymans
93d07298ff
session: let source keep track if it sent BYE
2013-07-26 12:17:56 +02:00
Wim Taymans
0c9c1434a8
source: reset more
2013-07-26 12:17:56 +02:00
Wim Taymans
1d02496d15
source: also use the source for bye_reason
...
Store the BYE reason in our internal source object. Rename the methods on the
source object a little because now the BYE can be received in RTCP or
set when the session wants to send BYE.
2013-07-26 12:17:56 +02:00
Wim Taymans
ddd071e54c
session: configure sdes with structure only
...
Remove code to configure the SDES with methods and types, only
allow configuration with GstStructure
2013-07-26 12:17:55 +02:00
Wim Taymans
0060e1d45d
session: refactor add and find source
...
Make functions to find and add a source to the hashtable.
2013-07-26 12:17:55 +02:00
Wim Taymans
adb0d68c07
session: remove source from sync_rtcp
...
We don't need to know the sender source of the session in the
callback, the SR packet is for all participants in the session.
2013-07-26 12:17:55 +02:00
Wim Taymans
bf7d8173b3
jitterbuffer: add some more debug
2013-07-26 12:17:55 +02:00
Wim Taymans
c44a29bd53
bin: fix compilation
2013-07-24 14:17:45 +02:00
Wim Taymans
f87875e35b
rtpjitterbuffer: fix locking
...
Take the lock earlier so that we do things that follow with the right
locking.
2013-07-24 10:49:03 +02:00
Wim Taymans
dece8413ef
rtpsession: don't use invalid times in RTCP timeouts
...
An invalid timeout can be calculated when we disabled RTCP by setting the
bandwidth to 0. Make sure all code can handle this case.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=674626
2013-07-23 17:41:48 +02:00
Wim Taymans
25e0f0d6b6
rtpsession: lock session when changing bandwidth
...
Take the session lock when changing the bandwidth properties so that we don't
end up with inconsistent behaviour.
2013-07-23 17:41:48 +02:00
Wim Taymans
c337265ee4
session: reset some RTCP variables
...
The early_send time was set to 0 and always triggering an early RTCP packet.
2013-07-23 17:41:48 +02:00
Carlos Rafael Giani
95429f1d4b
rtpbin: added custom downstream sync event
...
rtpbin can now send a custom in-band downstream event which informs
downstream that the bin has received an RTCP SR packet. This is useful
for applications which want to drop the initial unsynchronized received
RTP packets.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703560
Signed-off-by: Carlos Rafael Giani <dv@pseudoterminal.org>
2013-07-23 06:25:20 +02:00
Sebastian Dröge
5a9f4a3cbc
rtpsession: Add support for group-id in the stream-start event
2013-07-22 15:30:13 +02:00
Olivier Crête
3aa20e7c8d
rtpmux: Enable proxy caps on the src pads
2013-07-11 17:21:22 -04:00
Olivier Crête
1997acc8b2
rtpmux: Keep caps order from the peer or the filter
2013-07-09 17:43:31 -04:00
J. Rick Ramstetter
f01b751e52
rtp: Fix documentation and comments to use rtpbin instead of old gstrtpbin
...
https://bugzilla.gnome.org/show_bug.cgi?id=703426
2013-07-02 10:12:17 +02:00
Wim Taymans
519305d14d
jitterbuffer: improve sync on first packets
...
Don't throw away the first RTCP packet if it arrives before the first
RTP packet but remember and use it to signal sync once we get the
RTP packet.
See https://bugzilla.gnome.org/show_bug.cgi?id=691400
2013-06-27 16:23:20 +02:00
Wim Taymans
8969f00661
jitterbuffer: only signal loop when active
...
Only signal the loop function when it is active.
2013-06-27 16:15:45 +02:00
Wim Taymans
4bd2ffb26e
jitterbuffer: signal timestamp discont
...
We can now use the RESYNC buffer flag to mark a timestamp discont when we update
the ts-offset property.
2013-06-27 16:13:37 +02:00
Olivier Crête
2cd6f53e24
rtpptdemux: Wait after the caps to forward the other events
...
First forward the stream-start, then the caps, then the rest
2013-06-20 23:16:59 -04:00
Wim Taymans
51c9f7989f
rtpsession: Use the right hashtable to calculate bandwidth
...
Don't use an unused hashtable to iterate source to calculate bandwidth.
Remove unused code.
2013-06-13 16:02:19 +02:00
Wim Taymans
63f0ecbbe7
rtpsession: send stream-start and segment events
...
Also send stream-start and segment event on the RTCP pad.
We don't need to send anything on the sync_src pad because we
already forwarded all incomming events.
2013-05-28 12:26:25 +02:00
Nicolas Dufresne
04c9f43567
rtpmux: Send stream-start before caps
2013-05-13 15:37:05 +02:00
Sebastian Dröge
b0b0557c48
gst: Add better support for static plugins
2013-04-15 15:54:11 +02:00
Olivier Crête
6f3734c305
rtpssrcdemux: Only forward stick events while holding the sinkpad stream lock
...
Otherwise we get a race where if the RTCP packet comes in first and while
it is added the pads, the segment event arrives on the RTP stream, the event
may be lost completely and never forwarded.
2013-04-02 23:42:42 -04:00
Olivier Crête
76679f9ae9
rtpssrcdemux: No need to explicitely forward the caps
...
They are forwarded with the other events
2013-04-02 23:42:41 -04:00
Olivier Crête
4ad8693f3c
rtpssrcdemux: Remove unused GstSegment
2013-04-02 23:42:41 -04:00
Olivier Crête
7293b0eff7
rtpssrcdemux: Simplify event forwarding
...
Use the gst_pad_forward() mechanic, this way we won't miss pads that are
added while we are pushing
2013-04-02 23:42:41 -04:00
Olivier Crête
f4c3aef13a
rtpssrcdemux: Don't cross the internal links
...
We had the wrong condition to check for the internal links, so RTP and RTCP
pads got crossed!
2013-04-02 23:42:41 -04:00
Paul HENRYS
10802cae73
rtpsession: Fix wrong code organisation in case of collision
...
change_ssrc field of RTPSession should be set before calling
rtp_session_schedule_bye_locked () as this function will call reconsider function
that will wake up rtcp_thread which will call rtp_session_on_timeout () that will
check change_ssrc to change the ssrc.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=694184
2013-02-22 09:28:07 +02:00
Wim Taymans
a61055809f
rtpsession: only delay RTCP when we are a sender
...
Only delay the RTCP thread when we are a sender, which we can know because we
have a send_rtp_src pad. Otherwise we might delay the RTCP thread if we
are only a receiver and then there is no code path that wakes up the
RTCP thread and we end up without RTCP packets.
2013-02-20 21:07:41 +02:00
Wim Taymans
2d5319c1fa
rtpsession: delay RTCP until first RTP packet
...
Delay sending the first RTCP packet until we have sent the first RTP packet.
Otherwise we will send out a Receiver Report instead of a sender report.
See https://bugzilla.gnome.org/show_bug.cgi?id=691400
2013-02-08 17:05:27 +01:00
Wim Taymans
2971ed44ee
rtpsession: remove dead code
...
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=668355
2013-02-07 15:06:40 +01:00
Paul HENRYS
0e91c949d8
rtpptdemux: forward sticky events and then set caps
...
When a new src pad is added, first forward the sticky events and then
set the caps on the src pad
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692786
2013-02-07 14:38:20 +01:00
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
978cc9f538
rtpbin: rework cleanup of streams
...
Move the work of cleaning up the client streams in the free_stream
function. This allows us to properly clean up the client streams when we
remove an RTP stream as well.
Based on patch by Sujay <sdatar@cisco.com>
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=660156
2013-02-07 13:02:34 +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
Wim Taymans
747447d298
rtpsession: avoid '...is used uninitialized'
2013-01-29 10:32:51 +01:00
Tim-Philipp Müller
629772f735
rtpmanager: use C89-style comments
2013-01-28 23:07:34 +00:00
Olivier Crête
451217c437
gstrtpsession: Fix double-declared variable
2013-01-28 18:06:15 -05:00
Olivier Crête
7300d489fe
rtp: Fix compilation errors in previous patches
2013-01-28 17:58:20 -05:00
Haakon Sporsheim
86c13ceae6
rtpsession: Ensure MT safe event handling and plug event leak.
...
https://bugzilla.gnome.org/show_bug.cgi?id=667826
2013-01-28 17:44:31 -05:00
Idar Tollefsen
268c998a32
rtpsession: mt-safe event-push
...
By taking a ref of the sink-pad under lock, it won't dissappear
while the push is taking place
https://bugzilla.gnome.org/show_bug.cgi?id=667816
2013-01-28 17:34:50 -05:00
Pascal Buhler
f459fe2673
rtpssrcdemux: Safely push on pads that might be removed due to a RTCP BYE
...
https://bugzilla.gnome.org/show_bug.cgi?id=667815
2013-01-28 17:01:27 -05:00
Wim Taymans
72402cc649
rtp: small improvements
2013-01-08 16:27:42 +01:00
Wim Taymans
af055d9574
jitterbuffer: refactor handle sync code
...
Move the code that combines the last SR packet and the current jitterbuffer sync
values into a sync structure, into its own function. We want to reuse this bit
later.
2013-01-07 15:50:33 +01:00
Wim Taymans
87f7d6b9bf
rtp: include downstream latency in SR calculations
...
When we make a mapping between an RTP timestamp and an NTP timestamp, include
the downstream latency applied to the sinks. This makes it possible to have
both sinks run with different latencies and still have correct sync on the
client. It also is more correct because the RTP timestamp in the SR report will
actually correspond more closely to the NTP time it was sent on the server.
For pipelines with high latency on the sender side, this actually allows a
GStreamer receiver to perform synchronisation instead of dropping the RTCP
packets.
2013-01-07 15:45:10 +01:00
Wim Taymans
c631ed3300
rtpsession: don't cast event functions
...
There is no need to cast the event functions and only causes problems later when
we change the signature later and things silently compiles wrong code.
2013-01-07 14:25:14 +01:00
Wim Taymans
8dcde8b3ea
rtp: more debug
2013-01-07 14:23:34 +01:00
Wim Taymans
6b7d05ac57
rtpsession: improve debug
2013-01-07 14:22:48 +01:00
Tim-Philipp Müller
8a3b116d1f
docs: add rtpmux and rtpdtmfmux to plugin docs
...
https://bugzilla.gnome.org/show_bug.cgi?id=629117
2012-12-16 16:36:39 +00:00
Tim-Philipp Müller
3295b5d791
rtpmanager: move rtpmux and rtpdtmfmux elements from -bad
...
https://bugzilla.gnome.org/show_bug.cgi?id=629117
2012-12-16 16:36:39 +00:00
Tim-Philipp Müller
de204ba754
rtpmux: Fix FSF address
...
https://bugzilla.gnome.org/show_bug.cgi?id=687520
2012-12-16 16:36:39 +00:00
Tim-Philipp Müller
2778a1757f
rtpmux: Use gst_element_class_set_static_metadata()
...
where possible. Avoids some string copies. Also re-indent
some stuff. Also some indent fixes here and there.
2012-12-16 16:36:39 +00:00
Olivier Crête
15dfdc58d4
rtpmux: Misc fix for 0.11
...
Convert the incoming caps before proxying them
Clear the last_pad when going to ready
tests: Implement accept_caps, don't leak event
2012-12-16 16:36:38 +00:00
Wim Taymans
83262be703
rtpmux: update for RTP buffer api changes
2012-12-16 16:36:38 +00:00
Sebastian Dröge
f17064a8ea
rtpmux: Update for GST_PLUGIN_DEFINE() API changes
2012-12-16 16:36:34 +00:00
Wim Taymans
c86156ad8f
rtpmux: fix compilation
2012-12-16 16:35:36 +00:00
Wim Taymans
6826bbb6da
rtpmux: fix for caps api changes
2012-12-16 16:35:33 +00:00
Matej Knopp
bb345a584d
rtpmux: Fix compiler warnings
2012-12-16 16:35:29 +00:00
Olivier Crête
af4e999c59
rtpmux: Unref non-forwarded events
...
Also, don't unref forwarded ones
2012-12-16 16:35:29 +00:00
Olivier Crête
a8789d1df1
rtpmux: resync iterator on resync
2012-12-16 16:35:29 +00:00
Olivier Crête
0c54079af5
rtpmux: Re-push sticky events on input pad change
2012-12-16 16:35:29 +00:00
Olivier Crête
21831b430f
rtpmux: Don't leak gvalue from iterator
2012-12-16 16:35:29 +00:00
Wim Taymans
ccc4b960fc
rtpmux: more porting
2012-12-16 16:35:26 +00:00
Olivier Crête
f20a6b1d16
rtpmux: port to 0.11
2012-12-16 16:35:26 +00:00
Wim Taymans
35b6668fb6
rtpmux: make request pads take _%u
2012-12-16 16:35:22 +00:00
Olivier Crête
aa3607ef5c
rtpdtmfmux: Add last-stop to dtmf-event upstream events
...
Add the running time of the last outputted buffer to the
upstream "dtmf-event" events so that the dtmf source does not
leave a gap.
2012-12-16 16:35:22 +00:00
Edward Hervey
d137482fe5
rtpmux: Remove dead assignments
2012-12-16 16:35:22 +00:00
Stefan Kost
55aae6bfab
rtpmux: add missing G_PARAM_STATIC_STRINGS flags
...
Canonicalize property names as needed.
2012-12-16 16:35:15 +00:00
Olivier Crête
9674d5cc23
rtpmux: Improve documentation
...
Add an example pipeline, and try to explain a bit more what it does.
2012-12-16 16:35:15 +00:00
Stefan Kost
ca27a279ba
rtpdtmfmux: remove unused variable
2012-12-16 16:35:15 +00:00
Stefan Kost
c85dceeacb
rtpdtmfmux: remove unused signal boilerplate
2012-12-16 16:35:15 +00:00
Stefan Kost
2353f8d852
rtpmux: no need to ref pad in _chain()
2012-12-16 16:35:15 +00:00
Youness Alaoui
e42d2eebcb
rtpmux: Unlock the right mutex
...
The mutex locked is for the 'mux' object, but we unlock the
pad, which means that if the rtpmux gets a flush, then the
object lock will stay locked forever, causing it to freeze
the next time it tries to take it.
Fixes bug #627991
2012-12-16 16:35:15 +00:00
Olivier Crête
78d1ebac9e
rtpmux: Add support for GstBufferList
...
Factor out most of the buffer handling and implement a chain_list
function. Also, the DTMF muxer has been modified to just have a
function to accept or reject a buffer instead of having to subclass
both chain and chain_list.
2012-12-16 16:35:15 +00:00
Olivier Crête
c00f14419b
rtpmux: Don't leak invalid buffers
2012-12-16 16:35:15 +00:00
Tim-Philipp Müller
a45429d81d
rtpmux: fix missing debug log message argument
2012-12-16 16:35:15 +00:00
Olivier Crête
4a8d0243b5
rtpdtmfmux: Add some debug messages
2012-12-16 16:35:14 +00:00
Olivier Crête
423ce98666
rtpdtmfmux: Remove stream-lock event handling
2012-12-16 16:35:14 +00:00
Olivier Crête
a4500c0e74
rtpdtmfmux: Update doc for simplification
2012-12-16 16:35:14 +00:00
Olivier Crête
70097866de
rtpdtmfmux: Drop buffers on non-priority sinks when something is incoming on the priority sink
2012-12-16 16:35:14 +00:00
Olivier Crête
f6548fe9b6
rtpdtmfmux: Add priority sink pads
2012-12-16 16:35:14 +00:00
Olivier Crête
2bcea1537b
rtpdtmfmux: Cleanup event function
2012-12-16 16:35:14 +00:00
Olivier Crête
8e58646f5c
rtpmux: Aggregate incoming segments
2012-12-16 16:35:14 +00:00
Olivier Crête
7be57cac3a
rtpdtmfmux: Update documentation
2012-12-16 16:35:14 +00:00
Olivier Crête
e590fc1f32
rtpmux: Simplify request pad creation
2012-12-16 16:35:14 +00:00
Benjamin Otte
2867e00225
rtpmux: gst_element_class_set_details => gst_element_class_set_details_simple
2012-12-16 16:35:10 +00:00
unknown
fb7266884d
rtpmux: update the current_ssrc from the caps
...
Fixes #604101
2012-12-16 16:33:47 +00:00
Håvard Graff
eab65e84ca
rtpmux: release pads when disposing
...
Because of an allocated priv (GstRTPMuxPadPrivate), the element will
leak memory if not gst_rtp_mux_release_pad() is called. This would
previously only happen if release_request_pad() was called explicitly,
somthing that should not be neccesary.
Fixes #604099
2012-12-16 16:33:46 +00:00
Wim Taymans
0d54122804
dtmfmux: method name cleanups
2012-12-16 16:33:46 +00:00
Olivier Crête
3841cc74cf
rtpmux: Don't ignore requested pad name
2012-12-16 16:33:46 +00:00
Olivier Crête
d93295ff9d
rtpmux: Remove empty finalize
2012-12-16 16:33:46 +00:00
Olivier Crête
5e90a4e86b
rtpmux: Free the pad private data on pad release
...
Free the pad private data on pad release instead of using a weak ref,
which is not thread safe. Also, lock the content of the pad private using the element's
object lock.
2012-12-16 16:33:46 +00:00
Olivier Crête
4be63c9add
rtpmux: Reject wrong caps
2012-12-16 16:33:46 +00:00
Olivier Crête
0111bafb3a
rtpmux: Fix leak Fixed a leak discovered by Laurent Glayal <spegle@yahoo.fr>
2012-12-16 16:33:46 +00:00
Olivier Crête
fcc1522d2e
rtpmux: Fix leak
...
Fixed a leak discovered by Laurent Glayal <spegle@yahoo.fr>
2012-12-16 16:33:46 +00:00
Olivier Crête
ff6686f1c7
rtpmux: Fix warning
2012-12-16 16:33:46 +00:00
Olivier Crête
00791f930b
rtpmux: Set different caps depending on the input
2012-12-16 16:33:46 +00:00
Olivier Crête
ed0b407038
rtpmux: Only free pad private when pad is disposed
2012-12-16 16:33:45 +00:00
Olivier Crête
92bb5199ac
rtpmux: Remove useless caps mangling
2012-12-16 16:33:45 +00:00
Olivier Crête
3ccf3217fe
rtpmux: Rename variable for more clarity
2012-12-16 16:33:45 +00:00
Olivier Crête
4b958f6d8d
rtpmux: Use GST_BOILERPLATE
2012-12-16 16:33:45 +00:00
Olivier Crête
abe57be248
rtpmux: Do the includes locally
2012-12-16 16:33:45 +00:00
Olivier Crête
05844c89e9
rtpmux: Add GST_DEBUG_FUNCPTRs
2012-12-16 16:33:45 +00:00
Olivier Crête
fd102b95ab
rtpdtmfmux: Release locked pad on release_pad
...
Release the special pad if the pad is removed from the muxer.
2012-12-16 16:33:45 +00:00
Laurent Glayal
00f8bab712
rtpdtmfmux: Release special on pad dispose
...
Fixes #577690
2012-12-16 16:33:45 +00:00
Stefan Kost
a4a22454dc
docs: various doc fixes
...
No short-desc as we have them in the element details.
Also keep things (Makefile.am and sections.txt) sorted.
Reword ambigous returns. No text after since please.
2012-12-16 16:33:41 +00:00
Olivier Crête
7d4395a910
rtpmux: Move rtpmux from gst-plugins-farsight to -bad
2012-12-16 16:33:27 +00:00
Olivier Crête
68215752f4
rtpmux: Re-indent to Gst style
2012-12-16 16:33:24 +00:00
Olivier Crête
c7d0809434
rtpmux: Document rtp muxer a bit
2012-12-16 16:33:20 +00:00
Laurent Glayal
47c7a93df2
rtpmux: Add signals before stream lock and after unlocking
2012-12-16 16:33:17 +00:00
Olivier Crête
f1656ed8b0
rtpmux: Let ssrc through getcaps
2012-12-16 16:33:14 +00:00
Olivier Crête
1529dffaf9
rtpmux: Rename have_base to have_ts_base
2012-12-16 16:33:11 +00:00
Olivier Crête
57563517bd
rtpmux: Protect the seqnum with object lock in rtpmux
2012-12-16 16:33:08 +00:00
Olivier Crête
d3237eaf95
rtpmux: Remove unused sink_ts_base
2012-12-16 16:33:04 +00:00
Olivier Crête
cc23958183
rtpmux: Have getcaps to force the same clockrate on all pads
2012-12-16 16:33:01 +00:00
Olivier Crête
dc36590d0c
rtpmux: Validate RTP data in RTP Mux
2012-12-16 16:32:57 +00:00
Olivier Crête
360c8d4f1d
rtpmux: Remove unused clock-rate property
2012-12-16 16:32:54 +00:00
Olivier Crête
b86232d0dc
rtpmux: Clarify locking in rtpdtmfmux
2012-12-16 16:32:50 +00:00
Laurent Glayal
4b607cdda5
rtpmux: Missing format parameter
2012-12-16 16:32:47 +00:00
Håvard Graff
b313c80367
rtpmux: Update seqnum base in rtp muxer
...
With help from Wim
2012-12-16 16:32:43 +00:00
Håvard Graff
c479f90274
rtpmux: Fix some more leaks
2012-12-16 16:32:40 +00:00
Håvard Graff
1b5e769e0b
rtpmux: Fix leak
2012-12-16 16:32:37 +00:00
Olivier Crête
5cbb0de823
rtpmux: Don't unref caps we don't know (thanks Wim)
2012-12-16 16:32:32 +00:00
Olivier Crête
cebf506949
rtpmux: Put per-buffer debug at level LOG
2012-12-16 16:32:29 +00:00
Olivier Crête
3c12a423b7
rtpmux: Make debug print accurate
2012-12-16 16:32:25 +00:00
Olivier Crête
c49f4c87c6
rtpmux: Set our caps on the buffers
2012-12-16 16:32:22 +00:00
Olivier Crête
ec63da9366
rtpmux: Take the clock-base stored from the last setcaps
2012-12-16 16:32:18 +00:00
Olivier Crête
674c074114
rtpmux: Store the clock-base on setcaps
2012-12-16 16:32:15 +00:00
Olivier Crête
90264b9686
rtpmux: Add padprivate to the request pads
2012-12-16 16:32:11 +00:00
Olivier Crête
15d661ba3e
rtpmux: Make indentation more correct
2012-12-16 16:31:56 +00:00
Olivier Crête
3a7d09a749
rtpmux: Fix typo
2012-12-16 16:31:53 +00:00
Olivier Crête
91aef3ec5e
rtpmux: Set seqnum-base and clock-base in caps from rtpmuxer
2012-12-16 16:31:50 +00:00
Zeeshan Ali
6ea5ca354d
rtpmux: more debug
...
20070815135038-f3f1e-9c7a5490a525c6e8753cb1b8c03354df99132b5c.gz
2012-12-16 16:31:46 +00:00
Youness Alaoui
f0e209b638
rtpmux: missing comment
...
20070820185032-4f0f6-0ab67b6ac40dd4e35a8fe53f3cb6daff65ce43b9.gz
2012-12-16 16:30:33 +00:00
Olivier Crete
3ed5590da6
rtpmux: Make buffer writable before writing into it
...
20070712195336-3e2dc-91a5fb797cfa4919d4e2f9a728c6d6fbd3b83d93.gz
2012-12-16 16:30:31 +00:00
Olivier Crete
dd13f7c8ef
rtpmux: Set pads active when adding them to a potentially running element
...
20070706202459-3e2dc-a3731f885725594def0a7be997fc7b3a739ee967.gz
2012-12-16 16:30:27 +00:00
Olivier Crete
1c5075f927
rtpmux: Fix multiple ref leaks (patches by SP GLE)
...
20070607120121-3e2dc-061e9ef7a47b1b84fa8f8092f4b8bcc0e6db8c8c.gz
2012-12-16 16:30:23 +00:00
Zeeshan Ali
42f455e902
rtpmux: send event to all src pads
...
20070528152505-f3f1e-039216c73dc93f64c49962c77a0253cb9cfec4d3.gz
2012-12-16 16:30:18 +00:00
Zeeshan Ali
dba101bb0f
rtpmux: print a warning if receive an error iterating sinkpads
...
20070528123749-f3f1e-4c1eb3f511b5610143610a65a94d117f2c3d2580.gz
2012-12-16 16:30:15 +00:00
Zeeshan Ali
baa48dc6bc
rtpmux: deal with all the gst_iterator_next() return values
...
20070528122808-f3f1e-d301644c3be7633ec6dc5e28596e9346d2da6a50.gz
2012-12-16 16:30:12 +00:00
Zeeshan Ali
de40874670
rtpmux: Return correct value from the event handler
...
20070525123116-f3f1e-131b37b5f4521618fe2f1320409a47e65b35ad2d.gz
2012-12-16 16:30:08 +00:00
Zeeshan Ali
ed76f67e96
rtpmux: Ville's original patch to fix the traversal of dtmf event
...
20070525102709-f3f1e-6c41d1ef934068a4f4e810e7e981b420075b0c98.gz
2012-12-16 16:30:05 +00:00
zeeshan.ali@nokia.com
94ebe07862
rtpmux: Set the correct ts-offset on the get_prop value
...
20070329135250-65035-a43e222d91d57c0a61cb3287586aaa29abf78674.gz
2012-12-16 16:30:01 +00:00
zeeshan.ali@nokia.com
1ee542c378
rtpmux: Refactorize state_change
...
20070329135223-65035-23a0107b2e397710f035c6e88cc0e49b65bb4d5d.gz
2012-12-16 16:29:58 +00:00
zeeshan.ali@nokia.com
2498ba671a
rtpmux: set SSRC on the packets
...
20070329133622-65035-1be6e0aa85a71389f7d257b9cd3e13a73d6b745b.gz
2012-12-16 16:29:55 +00:00
zeeshan.ali@nokia.com
ee69c2690d
rtpmux: Code clean-up and more debug output
...
20070329131936-65035-9d499e209e0d7a409c3aa0d1040778babf076179.gz
2012-12-16 16:29:52 +00:00
zeeshan.ali@nokia.com
1c799ce964
rtpmux: Use own clock-base
...
20070328112219-65035-1ba5fefbc65059e9b0c860528a31062ceb6a7331.gz
2012-12-16 16:29:48 +00:00
zeeshan.ali@nokia.com
b04630d7a2
rtpmux: Only accept RTP streams that have the same clock-rate
...
20070323163139-65035-fc0b17b0b8a7a041f48994c4f26e96568168bf95.gz
2012-12-16 16:29:45 +00:00
zeeshan.ali@nokia.com
6fe1e02efd
rtpmux: Some more code-cleanups
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20070322161552-65035-bda96165e146b4f1d5fea1cc9576a7ab3abebc9e.gz
2012-12-16 16:29:42 +00:00
zeeshan.ali@nokia.com
1603223ee5
rtpmux: return newpad instead of NULL and warn if failed to create a pad
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20070322154251-65035-cdb6651e61c2eb0205cc8c24693b43f98a2da718.gz
2012-12-16 16:29:38 +00:00
zeeshan.ali@nokia.com
23d3ed5c5f
rtpmux: Refactorize the RTPMux code
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20070322124132-65035-0a3278147546e33f687097a43b775b3f6aa99f93.gz
2012-12-16 16:29:35 +00:00
zeeshan.ali@nokia.com
21e6e951f6
rtpmux: Some more doc fixing
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20070322121453-65035-12d602272217b51bd97df4e5790024c399622dd3.gz
2012-12-16 16:29:32 +00:00
zeeshan.ali@nokia.com
0de7fb6f37
rtpmux: More Refactoring
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20070322113228-65035-bae34a79599e7de5293ed77b022361ccff822bb9.gz
2012-12-16 16:29:29 +00:00
zeeshan.ali@nokia.com
0f755657ce
rtpmux: More documentation
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20070322113154-65035-624850541a5b5fc3df231204be5a83d07239db28.gz
2012-12-16 16:29:26 +00:00
zeeshan.ali@nokia.com
5483c78ac0
rtpmux: Refactor the event handler function
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20070321163311-65035-987e7f25d1ab5335b79f44b277abf15e4e37d317.gz
2012-12-16 16:29:23 +00:00
zeeshan.ali@nokia.com
db1523ae60
rtpmux: Add RTPDTMFMux element
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20070321145244-65035-9a01390b0dee3398e53199a1fa1d9352004f338e.gz
2012-12-16 16:29:19 +00:00
zeeshan.ali@nokia.com
97ff54dce7
rtpmux: Remove DTMF-specific code from RTP muxer and make it extendable
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20070321123149-65035-b8a8f55ff78eed8cbb0042e827885edfc5438242.gz
2012-12-16 16:29:16 +00:00
zeeshan.ali@nokia.com
1a227ac7e5
rtpmux: Put more helpful description
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20070320120524-65035-db27a7cf6307b511aeb3d996d26e790e367a7bad.gz
2012-12-16 16:29:13 +00:00
zeeshan.ali@nokia.com
d876c0d8cc
rtpmux: remove the (commented-out) code for blocking the pads
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20070316151641-65035-0123af387951f88594797c722e882cfe70240aff.gz
2012-12-16 16:29:10 +00:00
zeeshan.ali@nokia.com
209228c44d
rtpmux: Drop buffers instead of blocking the sinkpads
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20070316131444-65035-9c1345ad96108881f455d4b55a7f623cd302d0ed.gz
2012-12-16 16:29:05 +00:00
zeeshan.ali@nokia.com
795822ffa5
rtpmux: Implement stream locking, needed for DTMF
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20070314171618-65035-e4d24b1606ce0a3e2e739f01833f61e4d7555eac.gz
2012-12-16 16:29:02 +00:00
zeeshan.ali@nokia.com
fd209faa56
rtpmux: use GST_*_OBJECT instead of g_*
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20070314102058-65035-e2442888f2e3e5a3a7659ad7954a4fba34749ce2.gz
2012-12-16 16:28:58 +00:00
zeeshan.ali@nokia.com
b0208cb0a6
rtpmux: No need to manage pads, parent does that for us
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20070314101854-65035-ef5f4abde227102a1128835ab325905eae4c3726.gz
2012-12-16 16:28:55 +00:00
zeenix@gmail.com
74e9071dad
rtpmux: Fix copyright header
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20070314090358-d014a-3a6d3eeeaaf5cb8ca3bca6a33e99a551f598bd48.gz
2012-12-16 16:28:51 +00:00
zeeshan.ali@nokia.com
3c4cdf1541
rtpmux: The first implementation of RTP muxer
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20070307085307-65035-833402413f99cb3f8be4883e92bad4c8722510c9.gz
2012-12-16 16:28:41 +00:00
Havard Graff
9c94f1187c
jitterbuffer: bundle together late lost-events
...
The scenario where you have a gap in a steady flow of packets of
say 10 seconds (500 packets of with duration of 20ms), the jitterbuffer
will idle up until it receives the first buffer after the gap, but will
then go on to produce 499 lost-events, to "cover up" the gap.
Now this is obviously wrong, since the last possible time for the earliest
lost-events to be played out has obviously expired, but the fact that
the jitterbuffer has a "length", represented with its own latency combined
with the total latency downstream, allows for covering up at least some
of this gap.
So in the case of the "length" being 200ms, while having received packet
500, the jitterbuffer should still create a timeout for packet 491, which
will have its time expire at 10,02 seconds, specially since it might
actually arrive in time! But obviously, waiting for packet 100, that had
its time expire at 2 seconds, (remembering that the current time is 10)
is useless...
The patch will create one "big" lost-event for the first 490 packets,
and then go on to create single ones if they can reach their
playout deadline.
See https://bugzilla.gnome.org/show_bug.cgi?id=667838
2012-12-13 12:00:43 +01:00
Wim Taymans
c755af0cb0
rtpsource: protect against invalid RTP packets
2012-11-12 11:18:30 +01:00
Tim-Philipp Müller
230cf41cc9
Fix FSF address
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https://bugzilla.gnome.org/show_bug.cgi?id=687520
2012-11-04 00:07:18 +00:00
Wim Taymans
d5fd524a0c
rtsession: fix compiler warning
2012-10-17 13:55:45 +02:00
Wim Taymans
26a21e85e2
rtpbin: clarify the ntp-sync option
2012-10-17 13:35:07 +02:00
Wim Taymans
f17db5c4ed
rtpsession: update caps in the source
...
Inform the source when caps changed. This was removed in the port to 1.0
leaving the source unaware of the clock-rate and unable to interpollate
rtp timestamps for SR packets.
2012-10-17 13:22:40 +02:00
Wim Taymans
f4eef3f48d
rtpbin: set PTS and DTS in jitterbufffer
2012-10-17 12:46:32 +02:00
Wim Taymans
796c1d8029
rtpbin: disable check for ntp-sync
...
Disable the check for the ntp-sync method. It is expected that
a rather larger offset needs to be applied with this method.
2012-10-17 12:27:03 +02:00
Wim Taymans
1cebcfa8c2
rtpbin: use running-time for NTP time
...
When use-pipeline-clock is set, use the running-time of the
pipeline to calculate the NTP timestamps. This method would previously
only work when the base-time is set to 0 but with this change it can
also work with different offsets and we can also implement pause/resume
of the sender and receiver now.
2012-10-17 12:26:05 +02:00
Wim Taymans
5b394385b9
session: also stop probatation on existing sources
...
Receiving an RTCP packet should also stop probation on sources we have seen
before.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=683065
2012-08-30 22:07:24 +02:00
Aleix Conchillo Flaque
4a200b670f
rtp: make rtp packet probation configurable (bug #682512 )
2012-08-30 21:49:57 +02:00
Tim-Philipp Müller
4bb52bbadf
docs: gst-launch -> gst-launch-1.0 and ffmpegcolorspace -> videoconvert
2012-08-27 21:20:30 +01:00
Aleix Conchillo Flaque
8d864dbbfc
rtspsrc: make jitterbuffer drop-on-latency available ( fix #682055 )
...
Conflicts:
gst/rtsp/gstrtspsrc.h
2012-08-22 10:39:19 +02:00
Tim-Philipp Müller
787c314ec3
Silence some 'variable may be used uninitialized' compiler warnings
...
When compiling with -DG_DISABLE_ASSERT
2012-08-08 11:31:59 +01:00
Olivier Crête
2e21ace12c
rtpssrcdemux: Block pad while it is announced.
...
Block the RTP pad and associated RTCP pads while they are being
announced. This it to prevent a race where one is announced and
before the callback has connected it, the other one gets a buffer.
We can't use the "padlock" of ssrcdemux because it causes deadlocks.
2012-08-06 18:04:58 -07:00
Olivier Crête
2aa360c936
rtpssrcdemux: Release lock before signalling new pad
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This prevents a deadlock where something would try to push an event
through the SSRC demux from the callback, causing the pads to be iterated
and the lock taken.
2012-08-04 18:14:28 -07:00
Wim Taymans
51371d26ee
update for RTP buffer api changes
2012-07-17 16:38:27 +02:00
Tim-Philipp Müller
c22268b5d3
rtpsession: remove deprecated and unused "ntp-ns-base" property
2012-07-06 13:16:00 +01:00
Wim Taymans
30d3dfee36
update for task api change
2012-06-20 10:33:42 +02:00
Wim Taymans
dc04908412
update for clock api changes
2012-06-20 10:01:57 +02:00
Pascal Buhler
8161daef4a
rtpsession: creation should be signaled before validation
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https://bugzilla.gnome.org/show_bug.cgi?id=667850
2012-05-09 10:36:18 +02:00
Tim-Philipp Müller
e09ae5736d
Use new gst_element_class_set_static_metadata()
2012-04-10 00:51:41 +01:00