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525 commits

Author SHA1 Message Date
Wim Taymans
a4fdb8ee44 Merge branch 'master' into 0.11 
Conflicts:
	tests/examples/cairo/Makefile.am
 2011-03-08 10:14:20 +00:00
Mark Nauwelaerts
1f7f434df6 jitterbuffer: also estimate eos if very near eos 2011-03-07 16:56:43 +01:00
Mark Nauwelaerts
3c9a4239bf jitterbuffer: avoid trying to buffer more than is available. 
That is, in case of short (or near eos of) stream, deadlock (until timeout)
would occur trying to buffer more than is yet forthcoming.
 2011-03-07 16:56:18 +01:00
Mark Nauwelaerts
f84b8a69cb jitterbuffer: reset element base_time upon flush 
... to arrange for properly scheduled timeout (following seek).
 2011-03-07 11:07:12 +01:00
Wim Taymans
0a56b25882 rtpsession: use NetAddress metadata 2011-02-28 13:28:29 +01:00
Wim Taymans
d87c27fd2c miniobject: use buffer private field for extra data 
Use the owner private field to store extra buffer data instead of using
subclassing.
 2011-02-28 11:58:48 +01:00
Blaise Gassend
0f88181f43 rtpbin: handle NULL demux elements 
When using gstrtpbin with ignore-pt=true, the free_stream function tries to
call gst_element_set_locked_state and gst_element_set_state on a stream->demux
which is NULL.

fixes #642412
 2011-02-22 13:31:35 +01:00
Wim Taymans
45ea930a99 rtpbin: fix setting the SDES property 
Only the sdes veriable is protected with the object lock.
Use the right object when setting the sdes property.
 2011-02-21 17:19:05 +01:00
Wim Taymans
61382aad28 source: fix type of ntpnstime 2011-02-02 18:30:47 +01:00
Wim Taymans
8598aaf81b rtpbin: Get and use the NTP time when receiving RTCP 
When we receive an RTCP packet, get the current NTP time in nanseconds so that
we can correctly calculate the round-trip time.
 2011-02-02 18:30:46 +01:00
Olivier Crête
cd923223dd rtpsession: Add action signal to request early RTCP 2011-02-01 18:28:51 +01:00
Olivier Crête
c0996e6b90 rtpsession: Add callback to get the current time 2011-02-01 18:28:51 +01:00
Olivier Crête
a630c68fc3 rtpsession: Don't relay more than one PLI request per RTT 
Drop PLI requests if one was relay in the last RTT, the other side may
just not have received the keyframe yet.
 2011-02-01 18:28:51 +01:00
Olivier Crête
a61bb9e94b rtpsession: Send GstForceKeyUnit event in response to received RTCP PLI 2011-02-01 18:28:51 +01:00
Sjoerd Simons
7350d2adfa gstrtpsession: Fallback for FIR to PLI if PLI isn't available 2011-02-01 18:28:51 +01:00
Olivier Crête
52f95fa7ee rtpsession: Implement sending PLI packets in response to GstForceKeyUnit 2011-02-01 18:28:51 +01:00
Olivier Crête
db5150a23a rtpsource: Retain RTCP Feedback packets for a specified amount of time 2011-02-01 18:28:51 +01:00
Olivier Crête
90354ecb49 rtpsession: Make rtcp buffer metadata writable after processing it 
Functions that process the rtcp buffer could decide to keep a ref
on the buffer for further processing. So make the metadata writable
only after they are done.
 2011-02-01 18:28:50 +01:00
Olivier Crête
1643f427db rtpsession: Emit signal on incoming RTCP FB packet 2011-02-01 18:28:50 +01:00
Wim Taymans
f399b6a641 rtpsession: fix compilation 2011-02-01 18:28:50 +01:00
Olivier Crête
1bde427250 rtpsession: Add method to request early RTCP packet 
Implement the early mode defined in RFC 4585. In this mode, RTCP feedback
packets are sent early to notifier.
 2011-02-01 17:03:39 +01:00
Olivier Crête
975e1fecb3 rtpsession: Add property for minimum interval between Regular RTCP messages 
This can be changed according to RFC 4585
 2011-02-01 16:56:15 +01:00
Olivier Crête
cdb5465741 rtpsession: Emit signal when sending a compound RTCP packet 
This allows users to add extra RTCP packets to the compound
RTCP packet.
 2011-02-01 16:50:58 +01:00
Olivier Crête
589b254ce5 rtpptdemux: Tag upstream custom events with payload type 2011-02-01 16:50:25 +01:00
Olivier Crete
c7b1ce7310 rtpssrcdemux: Tag upstream custom events with SSRC 2011-02-01 16:49:10 +01:00
Olivier Crête
9f073459e0 rtpsession: Emit "on-ssrc-validated" when validating by RTCP 
Emit "on-ssrc-validated" if the SSRC is validated by receiving
a RTCP SDES packet.
 2011-02-01 16:45:58 +01:00
Stefan Kost
9f34b89245 rtpjitterbuffer: don't divide by 0 2011-01-25 21:57:57 +02:00
Wim Taymans
b5647685c4 rtpsource: use the right variable 
Use the right variable for specifying that we sent a receiver report.
 2010-12-27 13:13:46 +01:00
Wim Taymans
7caad21a57 rtpsource: include last send RB block 
Only report RB values for non-internal sources.
Report not only the RB blocks we last received from but also the last RB
block we sent to a source.
 2010-12-23 13:58:30 +01:00
Wim Taymans
8fa5ddab9a rtpsession: remember last sent RB values. 2010-12-23 13:58:30 +01:00
Wim Taymans
6035ee08c0 rtpsource: include all stats and document 
Include all possible stats of a source in the stats structure because we might
be interested in what happened in the past.
Document the stats property and the fields.
 2010-12-23 13:58:30 +01:00
Wim Taymans
10a5a795ea rtpsession: also emit RTCP activity on SR 
Also emit RTCP activity signals when we receive an SR packet without RB blocks,
such as from a sender that is not receiving anything.
 2010-12-23 13:58:30 +01:00
Wim Taymans
1230258e6f docs: add some more gstrtpbin docs 2010-12-23 13:58:29 +01:00
Wim Taymans
2b53cbe923 rtpsession: unlock before emitting signals 2010-12-22 11:46:21 +01:00
Wim Taymans
eb6d552353 jitterbuffer: get better buffering level 
When the jitterbuffer contains -1 timestamps, make sure we still calculate the
buffer fill level by skipping the -1 buffers.
Try to be more resilient to weird input timestamps.
 2010-12-20 15:56:50 +01:00
Wim Taymans
6cb0efede4 jitterbuffer: provide a clock. 
since we are using the clock for sync, we need to also provide a clock for good
measure. The reason is that even if downstream elements provide a clock, we
don't want to have that clock selected because it might not be running yet.
 2010-12-20 11:13:09 +01:00
Wim Taymans
210f1c44c7 rtpbin: copy buffering stats 
when we create an aggregate buffering message, copy the buffering stats form the
last message. At least we get correct buffering mode then.
 2010-12-20 11:13:09 +01:00
Wim Taymans
0c3333da04 session: fix average RTCP packet size some more. 
Fix stupid error in averaging macro.
Include udp headers in packet length estimation.
 2010-12-14 18:12:43 +01:00
Wim Taymans
7ebd374766 rtpbin: correctly calculate RTCP packet size 2010-12-14 17:15:23 +01:00
Wim Taymans
ffc7cd9803 jitterbuffer: avoid leaking sink events 
Avoid leaking the newsegment event when it has the wrong format.
 2010-12-13 12:57:58 +01:00
Mark Nauwelaerts
46c91476eb rtpssrcdemux: do not hold custom PAD_LOCK when pushing downstream 2010-12-03 15:50:21 +01:00
Olivier Crête
077a61932a rtpbin: Use the right constant to define the "use-pipeline-clock" property 
The wrong #define was being used, now use the correct one.
 2010-10-14 17:41:30 -04:00
Stefan Kost
d8167e3071 various (gst): add a missing G_PARAM_STATIC_STRINGS flags 2010-10-13 18:00:28 +03:00
Tim-Philipp Müller
d65eb2b91a ext, gst: canonicalise property names where this wasn't the case 
ie. "foo_bar" -> "foo-bar"
 2010-10-12 16:04:21 +01:00
Vladimir Eremeev
8bf7381385 rtpjitterbuffer: improve article reference in comment block 
https://bugzilla.gnome.org/show_bug.cgi?id=631082
 2010-10-01 18:07:03 +01:00
Thijs Vermeir
2c2c90a723 rtpjitterbuffer: update link to documentation 2010-09-30 12:08:49 +02:00
Pascal Buhler
7a8c2a4b8a rtpmanager: packet lost should not be a warning. It happens all the time... 2010-09-24 16:00:03 +02:00
Pascal Buhler
ca6a512b5e rtpbin: Make cleaning up sources in rtp_session_on_timeout MT safe 
Using _foreach_remove on the hashtable, while releasing the lock protecting
that table inside the callback is not a good idea. The hashtable might
then change (a source removed or added) while signals like on_timeout
are being sent.

This solution makes a copy of the table, performs the _foreach without
actually removing any sources, but marks them for removal on a second
iteration with the real list, but this time not letting go of the lock.

Fixes #630452
 2010-09-24 15:38:00 +02:00
Pascal Buhler
bd8d80a8e4 rtpbin: Handle rysnc of iterator when looking for free pad name 
If a new pad was added while iterating then a pad could be
returned that was already in use.

Fixes #630451
 2010-09-24 14:10:26 +02:00
Wim Taymans
8337c89c74 rtpsession: fix compilation 2010-09-24 14:10:26 +02:00
Trond Andersen
800b4bdb26 rtpbin: Unlock before adding pad in new_payload_found 
Holding internal locks while potentially calling out is a source
of deadlocks, and in this case the application might subscribe to the
pad-added signal.

Fixes #630449
 2010-09-24 14:00:11 +02:00
Havard Graff
062568a9f5 rtpsession: relax third-party collision detection 
If the source has been inactive for some time, we assume that it has
simply changed its transport source address. Hence, there is no true
third-party collision - only a simulated one.

Fixes #630447
 2010-09-24 13:56:56 +02:00
Wim Taymans
ce007b244e rtpsource: whitespace fixes 2010-09-24 13:50:02 +02:00
Wim Taymans
c5203a479b rtpsource: simplify the rate estimation some more 2010-09-24 13:48:50 +02:00
Havard Graff
0fa589a3dd rtpmanager: provide additional statistics 2010-09-24 13:26:10 +02:00
Wim Taymans
2c8b725591 rtpstats: printf format fixes 2010-09-17 11:07:52 +02:00
Olivier Crête
8e73da10b3 gstrtpsession: Split getting the caps into its own function 2010-09-13 16:25:42 +02:00
Wim Taymans
8e1c9b5b33 rtpbin: small cleanup. 2010-09-13 16:25:42 +02:00
Wim Taymans
d541f5e24d rtpsession: Small cleanups 
Make the property description prettier.
Actually multiple the bandwidth with the fraction.
 2010-09-13 15:51:20 +02:00
Olivier Crête
1f17b334ff rtpsession: Calculate RTCP bandwidth as a fraction of the RTP bandwidth 
Calculate the RTCP bandwidth to be a fraction of the RTP bandwidth if it is
specified as a value between 0 and 1.
 2010-09-13 15:51:20 +02:00
Wim Taymans
8381d9788d session: improve bandwidth recalculation 
Also recalculate bandwidth when one of the source bandwidths changed.
Use the newly calculated bandwidth.
 2010-09-13 15:51:20 +02:00
Olivier Crête
6f53a2b240 rtpsession: Add the option to auto-discover the RTP bandwidth 2010-09-13 15:51:19 +02:00
Thijs Vermeir
f38e37470a rtpbin: set use-pipeline-clock on correct GObject 2010-09-13 14:39:51 +02:00
Olivier Crête
94e87ef8ee rtpsession: Initialise the average scaled by 16 2010-09-13 13:10:19 +02:00
Wim Taymans
e6db74764b rtpsession: add running_time argument docs 2010-09-13 12:41:56 +02:00
Olivier Crête
00fd89c074 rtpstats: Rectify description of current_time in RTPArrivalStats 
It is the current time, it is unrelated to when the packet was actually received.
 2010-09-13 12:37:01 +02:00
Wim Taymans
cb6de429a0 rtpsession: compute the average correctly scaled 2010-09-13 12:31:40 +02:00
Olivier Crête
64e4ffa25b rtpsession: Count sent RTCP packets after they have been finished 
If they are counted before calling gst_rtcp_buffer_end(), then the
size is way too big.
 2010-09-13 12:13:23 +02:00
Olivier Crête
306ee454c6 gstrtpsession: Don't unref pads in finalize 
The gstrtpsession object is not holding any reference to them directly
 2010-09-13 12:10:11 +02:00
Wim Taymans
93228ccd52 rtpbin: add ntp-sync property 
Add an ntp-sync property that will sync the received streams to the server
NTP time. This requires synchronized NTP times between the sender and receivers,
like with ntpd.

Based on patch from Thijs Vermeir.

Fixes #627796
 2010-09-06 11:01:57 +02:00
Wim Taymans
f03fd91400 jitterbuffer: rename a variable to avoid confusion 2010-09-06 11:01:57 +02:00
Wim Taymans
e3479630ae rtpbin: rename some variables for less confusion 2010-09-06 11:01:57 +02:00
Wim Taymans
0f59664c6a rtpjitterbuffer: move comment where it belongs 2010-09-06 11:01:57 +02:00
Wim Taymans
4fd81747f3 session: minor cleanups 
Make clock snapshots more accurate by only sampling the same clock once.
 2010-09-06 11:01:57 +02:00
Thijs Vermeir
51020549f0 rtpbin: add use-pipeline-clock property 
With this property RTCP SR NTP times can be based
on the system clock (maybe synced with ntpd) or the
current pipeline clock.

https://bugzilla.gnome.org/show_bug.cgi?id=627796
 2010-09-06 11:01:57 +02:00
Thijs Vermeir
244a35a226 rtpptdemux: fix memleak on custom downstream events 
by not sending custom downstream event twice and fix memleak when
not handling the event

https://bugzilla.gnome.org/show_bug.cgi?id=623196
 2010-06-30 12:39:09 +02:00
Sebastian Dröge
f16ed4a91c gst: Don't use GST_DEBUG_FUNCPTR for GObject vfuncs 2010-06-06 17:52:40 +02:00
Thijs Vermeir
0bb2be3a7e rtpjitterbuffer: fix compiler warning 
unused variable ‘estimated’
 2010-06-02 15:32:36 +02:00
Alessandro Decina
4b6cb93025 rtpjitterbuffer: stop buffering and emit EOS at the end of a stream 
When using RTP_JITTER_BUFFER_MODE_BUFFER, make sure that the ringbuffer doesn't
get stuck buffering forever when there isn't enough data left to fill the
buffer.
 2010-06-02 14:21:16 +02:00
Wim Taymans
dc2662e22b rtpbin: fix docs 
Documentation error spotted by tony <caicai0119 at gmail.com>

Fixes #618419
 2010-05-13 13:01:26 +02:00
Wim Taymans
50f26c671b rtpsession: fix return value 2010-05-07 19:06:35 +02:00
Wim Taymans
aadf4ddf7e rtpsession: add properties to configure the bandwidth 
Add properties to proxy the bandwidth configuration to the session object.
 2010-05-07 18:58:58 +02:00
Wim Taymans
69cde0e874 rtpsession: add properties to configure bandwidths 
Add properties to configure the sender and receiver bandwidths.
Configure the bandwidths before calculating the RTCP timeout when we need to.
 2010-05-07 18:57:13 +02:00
Wim Taymans
d84dc1112d rtpstats: add some debug info 2010-05-07 18:56:30 +02:00
Wim Taymans
5690331c9e rtpsession: small cleanups 2010-05-07 18:55:34 +02:00
Wim Taymans
0da5cf2e21 rtpstats: make bandwidths more configurable 
Add a method to configure the various bandwidths in the session.
 2010-05-07 16:55:13 +02:00
Wim Taymans
6eee730c4a rtpsession: handle NONE RTCP intervals 
Prepare for handling RTCP reporting intervals of GST_CLOCK_TIME_NONE, which
means don't send RTCP at all.
 2010-05-07 13:32:30 +02:00
Alessandro Decina
40899379c0 rtpjitterbuffer: move some initialization code from change_state to _init. 
Set ->active to TRUE in _init so it can be set to FALSE after creating the
jitterbuffer and it won't be mistakenly reset to TRUE in the change_state
function.
This is needed to start the jitterbuffer as inactive when rtpbin is buffering.
 2010-05-03 13:34:59 +02:00
Alessandro Decina
ffc2da30fc rtpbin: fix a bug handling BUFFERING messages. 
If a session exists but has no streams, set the min buffering percent to 0
since it means that we haven't received anything for that session yet.
 2010-05-03 11:56:58 +02:00
Alessandro Decina
f6e9f359b9 rtpbin: when a stream is created, pause the jitterbuffer if rtpbin is buffering. 2010-05-03 11:51:37 +02:00
Alessandro Decina
38a5b08ef2 rtpbin: fix a bug calculating stream offsets. 2010-05-03 11:23:59 +02:00
Stefan Kost
d6e9af2a11 docs: do proper escaping for "%" 2010-04-08 18:05:46 +03:00
Stefan Kost
9967a4112b rtpsession: remove prototype for non existing function 
There is no function by that name anywhere.
 2010-04-08 14:02:50 +03:00
Benjamin Otte
cccfeaa59c gst_element_class_set_details => gst_element_class_set_details_simple 2010-03-18 14:32:00 +01:00
Benjamin Otte
1055aaa9cb Add -Wredundant-decls warning flag 
Also fix compile issues
 2010-03-17 19:35:10 +01:00
Benjamin Otte
21f66635e8 Update for recent changes to common submodule 
This just replaces every "$ERROR_CFLAGS" usage with a usage of
"$WARNING_CFLAGS $ERROR_CFLAGS" to get the same functionality as
previously.

Actually using that separation will happen later.
 2010-03-10 21:53:51 +01:00
Olivier Crête
a6dfe96169 rtpsession: Make it possible to favor new sources in case of SSRC conflict 
Add a "favor-new" property that tells the session to favor new sources when
there is a SSRC conflict. This is useful for SIP calls and other such cases
where a remote loop is extremely unlikely.

Fixes #607615
 2010-03-10 11:21:19 +01:00
Olivier Crête
f336ea283f rtpsession: Move SSRC conflicts lists into RTPSource 
We will also need to track SSRC conflicts in remote sources.

See #607615
 2010-03-10 11:21:18 +01:00
Wim Taymans
529f443a61 rtpsource: use payload size to estimate bitrate 
Use the length of the payload for estimating the receiver bitrate so that it
matches the calculations done on the sender side. Together with the number of
packets one can scale the bitrate with the header overhead of the lower
transport.
 2010-03-08 17:48:04 +01:00
Wim Taymans
c971d1a9ab rtpsource: refactor bitrate estimation 
Don't reuse the same variable we need for stats for the bitrate estimation
because we're updating it.
Refactor the bitrate estimation code so that both sender and receivers use the
same code path.
 2010-03-08 17:48:00 +01:00
Tristan Matthews
a0a6d4ff3b added bitrate estimation to receiver-side stats, fixes #611213 2010-03-08 17:47:55 +01:00
Sebastian Dröge
bcd06ea527 rtpjitterbuffer: Reset skew detection after instantiating the jitterbuffer 
...not only when going to READY. This sets high_level and friends to
a more useful value.
 2010-02-23 17:24:03 +01:00
Sebastian Dröge
0a12e69024 rtpjitterbuffer: Return 100 if high-level is 0 instead of dividing by zero 2010-02-23 17:20:02 +01:00
Tim-Philipp Müller
07fa73f199 docs: add Since: markers for new jitterbuffer properties 2010-02-19 12:13:07 +00:00
Wim Taymans
9d40d60960 rtpbin: remove use of ntp_ns_base 2010-02-15 21:36:29 +01:00
Wim Taymans
5a4ecc9da1 rtpbin: remove more ntpnstime and cleanups 
Remove some code where we pass ntpnstime around, we can do most things with the
running_time just fine.
Rename a variable in the ArrivalStats struct so that it's clear that this is the
current system time.
 2010-02-15 21:36:29 +01:00
Wim Taymans
74241e549f rtpsource: use running_time for jitter 
Use the running_time to calculate the jitter instead of the ntp time. Part of
the plan to get rid of ntpnsbase.
 2010-02-15 21:36:29 +01:00
Wim Taymans
83cb1aecc8 rtpbin: change how NTP time is calculated in RTCP 
Don't calculate the NTP time based on the running_time of the pipeline but from
the systemclock. This allows us to generate more accurate NTP timestamps in case
the systemclock is synchronized with NTP or similar.
 2010-02-15 21:36:29 +01:00
Tim-Philipp Müller
63c86ac3d8 raw1394, matroska, rtpmanager: remove padding from structures 
None of these element and class structures are in public headers,
so don't need padding.
 2010-02-15 00:50:10 +00:00
Wim Taymans
7f08081016 jitterbuffer: don't resync to invalid timestamps 
If we detect backward timestamps on the server, don't try to resync when we
don't have an input timestamp (such as when using RTSP over TCP) instead, do
nothing but assume the timestamp was ok, it will correct itself when time goes
forwards.
 2010-02-12 19:32:27 +01:00
Wim Taymans
d344754f03 rtpbin: fix typo 2010-02-12 17:22:56 +01:00
Wim Taymans
772eca5aff jitterbuffer: start out active and not buffering 
There is no need to set the latency in the jittebuffer in _init, we will set
that later when going to PAUSED.
Set the jitterbuffer active and not buffering when starting.
 2010-02-12 17:22:56 +01:00
Wim Taymans
8bbfd94c25 rtpbin: more buffering work 
When deactivating jitterbuffers when the buffering starts, keep the current
percent of the jitterbuffer and also set the jitterbuffer in the buffering state
so that we know when it's filled again.
Add property to get the buffering percentage of the jitterbuffer.
 2010-02-12 17:22:56 +01:00
Wim Taymans
e6e287cdcc rtpjitterbuffer: adjust latency in buffer mode 
When we are in buffer mode, adjust the buffering low/high thresholds based on
the total configured latency. If we don't and there is a huge queue or element
with a big latency downstream we might drain the complete queue immediately and
start buffering again.
 2010-02-12 17:22:55 +01:00
Wim Taymans
ab73603031 jitterbuffer: add ts-offset to timestamp 
Add the ts-offset to the buffer timestamp to get the final output timestamp of
the buffer.
 2010-02-12 17:22:55 +01:00
Wim Taymans
74a3be350d rtpbin: do more accurate buffer offsets 
Return the next timestamp in the jitterbuffer.
Use the min-timestamp of the jitterbuffers to calculate an offset so that the
next timestamp is pushed with a timestamp equal to running_time.
Start producing timestamps from 0 in the buffering case too.
 2010-02-12 17:22:55 +01:00
Wim Taymans
3efcc0fbc1 rtpbin: only start buffering when < 100% 
Only start buffering when the percentage message is < 100 %.
 2010-02-12 17:22:55 +01:00
Wim Taymans
0348ebe651 rtpbin: keep track of elapsed pause time 
Keep track of the time we spend pausing the jitterbuffers when they were
buffering and distribute this elapsed time to the jitterbuffers.
Also keep the latency in nanosecond precision.
 2010-02-12 17:22:54 +01:00
Wim Taymans
ecf6ed8fc1 jitterbuffer: keep track of offset 
Keep track of an outgoing offset that we add to each outgoing buffer to
compensate for PAUSE when buffering.
Adjust the offset when activating.
 2010-02-12 17:22:54 +01:00
Wim Taymans
048e5b6fbe jitterbuffer: report level using high watermark 2010-02-12 17:22:54 +01:00
Wim Taymans
8d814f3782 rtpbin: pass running_time to jitterbuffer pause 
Pass the current running time to the jitterbuffer when pausing or resuming so
that it calculate the right offsets.
Small cleanups and comments.
Set the default rtspsrc latency to 2 seconds.
 2010-02-12 17:22:54 +01:00
Wim Taymans
bf697b12e3 rtpbin: add some comments 2010-02-12 17:22:53 +01:00
Wim Taymans
20a27a545a rtpbin: more buffering updates 
Add signal to pause the jitterbuffer. This will be emitted from gstrtpbin when
one of the jitterbuffers is buffering.
Make rtpbin collect the buffering messages and post a new buffering message with
the min value.
Remove the stats callback from jitterbuffer but pass a percent integer to
functions that affect the buffering state of the jitterbuffer. This allows us
then to post buffering messages from outside of the jitterbuffer lock.
 2010-02-12 17:22:53 +01:00
Wim Taymans
a5b9d3f917 rtpbin: propagate buffer-mode property 
Propagate buffer-mode property to the jitterbuffers.
Intercept BUFFERING messages in rtpbin
 2010-02-12 17:22:53 +01:00
Wim Taymans
d3db9574a9 jitterbuffer: do more buffering implementation 
Add callback for buffering stats.
Configure the latency in the jitterbuffer instead of passing it with _insert.
Calculate buffering levels when pushing and popping
Post buffering messages.
 2010-02-12 17:22:52 +01:00
Wim Taymans
aeacbfed3e jitterbuffer: flesh out buffering mode some more 
Add a buffering state to the jitterbuffer and wait until buffering ends before
pushing out packets.
 2010-02-12 17:22:52 +01:00
Wim Taymans
56b29c9a6b jitterbuffer: hook up the mode property 
Expose a mode property on the jitterbuffer.
Fix the case where timestamps are -1 in the check for outgoing timestamps.
 2010-02-12 17:22:52 +01:00
Wim Taymans
be4517a6b8 jitterbuffer: add buffering mode options 
Add getters and setters for different buffering modes that the jitterbuffer will
support. Default to the current slave mode.
 2010-02-12 17:22:52 +01:00
Wim Taymans
99a581215f jitterbuffer: add some more debug info 2010-02-12 13:53:57 +01:00
Wim Taymans
05418f1687 rtpbin: avoid some structure copies 
Don't make copied in the getter and setter for SDES in the RTPSource. This
avoids a couple of copies of the SDES structure when generating RTCP
packets.
 2009-12-22 22:27:21 +01:00
Pascal Buhler
c3448f978e rtpmanager: improve SDES handling 
Store SDES internally as a struct to support multiple PRIV values.
Include all values set in SDES struct when sending RTCP SDES.
 2009-12-22 21:43:25 +01:00
Wim Taymans
9734699788 rtpbin: add property to remove pads automatically 
Add a property called autoremove to automatically remove the pads of sources
that timed out.

Fixes #554839
 2009-12-21 15:07:44 +01:00
Wim Taymans
c611bbaa8e ssrcdemux: fix comparison 
A NULL means no pad was found.
 2009-12-21 15:07:34 +01:00
Aurelien Grimaud
07f27f0efd rtpsession: avoid buffer ref/unref pairs for CSRCs 
We ref the buffer before pushing it downstream in order to get the CSRCs of it
after pushing. This causes performance problems when downstream elements want to
change the metadata because the buffer needs to be subbuffered.

Instead, read and store the CSRCs of the buffer in an array before pushing it
and process the array after pushing the buffer. This allows us to remove the
ref/unref pair.

Fixes #603376
 2009-11-30 15:59:50 +01:00
Wim Taymans
8070ae967b jitterbuffer: avoid using wrong clock-rate 
Check for a valid clock-rate before attempting to estimate the npt
stop time.
 2009-11-25 10:38:23 -06:00
Wim Taymans
5682e2bf01 rtpbin: fix typo in comments 2009-11-25 10:37:30 -06:00
Stefan Kost
9ee0815e85 docs: more links and better short description 
Fix spelling of GstRtpSsrcDemux to get it linked. Add more links. Change
the short description to be more meaningful.
 2009-11-20 11:25:49 +02:00
Wim Taymans
f52859432f jitterbuffer: release lock before emiting signals 
Release the jbuf lock before emiting the request-pt-map signal to avoid
deadlocks. We also need to catch the shutdown case when locking again.

Fixes #593354
 2009-11-18 10:50:44 +01:00
Stefan Kost
e43eb89449 tests: add a jitterbuffer test 
Tests pushing a few buffers in various order and asserting the order sent by the
jitterbuffer. Contains two disabled tests that need more work.
 2009-10-22 13:35:57 +03:00
Stefan Kost
6904e46ef2 build: use gst-glib-gen.mak to fix the glib build rules. 
The build rules in glib-gen.mak were using pattern rules in a non save way.
 2009-10-16 11:53:38 +03:00
Håvard Graff
58b9de4cca rtpptdemux: only forward the lost-event to the last seen pt-number 
forward all events on all pads except for the PacketLost event, which we want to
forward to the last seen pt pad.

Fixes #598377
 2009-10-14 12:28:55 +02:00
Stefan Kost
e0cdd879b4 build: fprintf, sprintf, sscanf need stdio.h 2009-10-07 14:03:20 +03:00
Wim Taymans
0040d01265 rtpbin: use locking around the sessions 2009-10-05 16:07:24 +02:00
Wim Taymans
8fb77403c5 jitterbuffer: cache latency in nanoseconds 
Cache the latency in nanoseconds units to avoid having to convert the
milliseconds value to nanoseconds all the time.
 2009-10-01 12:52:40 +02:00
Wim Taymans
c262735164 jitterbuffer: handle -1 input timestamps 
Don't try to check a -1 timestamp against the max delay.
 2009-10-01 12:12:09 +02:00
Stefan Kost
0a7ef67ad0 docs: fix gtk-doc warnings 2009-09-10 10:28:48 +03:00
Marc Leeman
6b46aeb6a3 rtpbin: add ignore-pt parameter 
Add a parameter 'ignore-pt' that disables creating a gstrtpptdemux module and
ghosts the pads of gstrtpjitterbuffer instead of the ones of gstrtpptdemux.

Fixes #594490
 2009-09-08 17:38:32 +02:00
Håvard Graff
2912b21d14 rtpbin: propagate payload-type-change signal from demuxer 
fixes #594254
 2009-09-08 13:59:56 +02:00
Havard Graff
a52309eff7 jitterbuffer: change severity of clock-rate change debug 
Make log GST_DEBUG under normal circumstances, GST_WARNING otherwise.

Fixes #594253
 2009-09-08 13:44:49 +02:00
Håvard Graff
40549278c3 jitterbuffer: avoid throwing reordered buffers with same timestamps 
When we receive a reordered packet with the same timestamp as the previous one
(which can happen for fragmented packets) don't consider the packet as lost but
instead wait for the reordered packet to arrive.

Switch the warning-level, so that a reordering does not get a warning, only
an actual produced lost-packet.

Fixes #594251
 2009-09-08 13:39:31 +02:00
Stig Sandnes
8f3299c547 rtpbin: make free_session() remove stream references 
When receiving a sync-packet, all sessions with the same cname will be compared
and synced together. In this process, there could still be references to a
session that has been shut down in the meanwhile.

This patch makes sure that these references are removed when shutting down a
session, so that the syncing can be done safely.

Fixes #594283
 2009-09-08 13:18:29 +02:00
Havard Graff
e08e610db0 rtpbin: use locked state on internal bins 
Set the locked state on internal elements to make sure that they don't change
back to another state when shutting down.

Fixes #594248
 2009-09-08 12:41:52 +02:00
Laurent Glayal
371875c57a rtpsource: fix memleak 
Don't leak the input buffer when the received and expected seqnum are different when
in probation.

fixes #594039
 2009-09-03 19:37:10 +02:00
Olivier Crête
f542f710cf rtpjitterbuffer: Lock clock_rate variable 
The priv->clock_rate variable could become -1 between when its checked to not
be -1 and when its used, causing an assertion. Fixed by taking the mutex
earlier in the chain() function.

Fixes #593955
 2009-09-03 19:17:00 +02:00
Wim Taymans
3fcde4486d rtpsource: whitespace fixes 2009-09-03 19:17:00 +02:00
Wim Taymans
3f629f6001 rtpsession: whitespace fixes 2009-09-03 19:16:59 +02:00
Peter Kjellerstedt
fdf18653b7 rtpmanager: Fixed a copy & paste error 2009-09-01 15:06:46 +02:00
Peter Kjellerstedt
dc4f9575be rtpmanager: Removed unused variable priv 
The variable priv was initialized in a lot of functions but then never
used for anything.
 2009-09-01 13:21:23 +02:00
Peter Kjellerstedt
57adc2a803 rtpmanager: A little clean up 
Make the code flow of gst_rtp_session_send_rtcp() and
gst_rtp_session_sync_rtcp() identical.
 2009-09-01 13:04:14 +02:00
Peter Kjellerstedt
923b5b495a rtpmanager: Make sure that used caps are not freed already (take 2) 
This reintroduces the fix for bug #593391. It also applies it in
gst_rtp_session_sync_rtcp() which has very similar code to
gst_rtp_session_send_rtcp().
 2009-09-01 13:04:14 +02:00
Wim Taymans
8d924611e7 jitterbuffer: make sure time does not go backwards 
When we construct a timestamp that would result in a timestamp that is earlier
than when the packet was received, reset the skew calculation as this is
probably a sign that the sender restarted or paused.

Fixes #593354
 2009-09-01 12:48:28 +02:00
Peter Kjellerstedt
bfb1260af4 rtpmanager: Set caps in gst_rtp_session_send_rtcp() correctly again 
The test for when to set an RTCP caps on the output pad in
gst_rtp_session_send_rtcp() accidentally got inverted in the last commit.
 2009-09-01 11:32:41 +02:00
Wim Taymans
a74c385b7b rtpsession: use proper locking for pads and caps 
Use the sesion lock and shotdown variable to protect and ref the pads we are
going to push on.

fixes #561825
 2009-08-31 16:38:27 +02:00
Wim Taymans
a522a2d4d2 rtpbin: whitespace fixes 2009-08-31 16:33:26 +02:00
Wim Taymans
a26a2a9ff5 jitterbuffer: add slope estimation code and debug 
Add some code to measure the sender speed vs the receiver speed. This can be
used to detect bursts.
 2009-08-31 13:02:16 +02:00
Wim Taymans
4814d899c2 jitterbuffer: reset skew when timestamps change 
Refactor the jitterbuffer resync code.
Reset the skew correction when we detect a big timestamp discont.

See #593354
 2009-08-31 12:57:32 +02:00
Wim Taymans
e254936e34 jitterbuffer: make sure time never goes invalid 
Since the skew can be negative, we might end up with invalid timestamps. Check
for negative results and clamp to 0.

See #593354
 2009-08-31 12:47:15 +02:00
Sebastian Dröge
041fa82179 rtpsession: Make sure that used caps are not freed already 
Fixes bug #593391.
 2009-08-31 08:09:09 +02:00
Sebastian Dröge
000a483d31 rtp: Use new gst_iterator_new_single() for the internal linked pads iteration 2009-08-31 08:09:09 +02:00
Sebastian Dröge
a1cddb3fd6 rtpsession: Use iterate internal links instead of deprecated get internal links 2009-08-31 08:09:09 +02:00
Sebastian Dröge
c8c02d2c7a jitterbuffer: Use iterate internal links instead of deprecated get internal links 2009-08-31 08:09:08 +02:00
Sebastian Dröge
97cb7bdb6c rtpssrcdemux: Use iterate internal links instead of deprecated get internal links 2009-08-31 08:09:08 +02:00
Olivier Crête
7f569ca9c8 rtpbin: Fix reference leak 
Fixes #591476.
 2009-08-14 13:47:18 +01:00
ric
92abe07e80 rtpsource: avoid buffer leak on bad seqnum 
Fixes #590797
 2009-08-11 02:30:47 +01:00
Wim Taymans
9f68303a2e rtpsource: allow for NULL caps on buffers 
Add the NULL caps check where it matters and also cover another case of
potential NULL caps.

Fixes #590030
 2009-08-11 02:30:47 +01:00
Olivier Crête
e37844fdc7 rtpsource: Incoming buffers do not always have caps 2009-08-11 02:30:47 +01:00
Wim Taymans
3091137217 rtpsession: avoid doing lip-sync in BYE 
When we get a BYE packet, don't do lip-sync with the SR inside because some
senders have trouble constructing valid SR packets after BYE.
 2009-08-11 02:30:47 +01:00
Wim Taymans
3747ede14a rtpbin: don't do lip-sync after a BYE 
After a BYE packet from a source, stop forwarding the SR packets for lip-sync
to rtpbin. Some senders don't update their SR packets correctly after sending a
BYE and then we break lip-sync. We prefer to let the jitterbuffers drain with
the current lip-sync instead.
 2009-08-11 02:30:47 +01:00
Wim Taymans
d2ef095b80 rtpbin: only reconsider once for BYE 
When iterating the sources of a BYE packet, don't signal a reconsideration for
each of them but signal after we handled all sources.
 2009-08-11 02:30:47 +01:00
Olivier Crête
e8c6bcdf8d rtpsession: Free conflicting addresses on finalize 2009-08-11 02:30:46 +01:00
Wim Taymans
428368b44a rtpbin: use new method for netaddress to string 2009-08-11 02:30:46 +01:00
Wim Taymans
512ba93159 rtpbin: do better cleanup of the src ghostpads 
Connect to the pad-removed signal of the ptdemux elements so that we remove the
ghostpads for them. Fixes cleanup when going to NULL as well as when releasing
the sinkpads.

Fixes #561752
 2009-08-11 02:30:46 +01:00
Wim Taymans
d7a8663e05 rtpsession: add a comment 2009-08-11 02:30:46 +01:00
Wim Taymans
c53e595d23 rtpbin: add SDES property 
Remove all individual SDES properties and use one sdes property that takes a
GstStructure instead. This will allow us to add more custom stuff to the SDES
messages later.
 2009-08-11 02:30:46 +01:00
Wim Taymans
9f330992f5 rtpbin: add SDES property that takes GstStructure 
Remove all individual SDES properties and use one sdes property that takes a
GstStructure instead. This will allow us to add more custom stuff to the SDES
messages later.
 2009-08-11 02:30:46 +01:00
Wim Taymans
d8496fb105 rtpbin: removed old gstrtpclient 2009-08-11 02:30:45 +01:00
Branko Subasic
779f67adc4 rtpbin: add support for buffer-list 
Add support for sending buffer-lists.
Add unit test for testing that the buffer-list passed through rtpbin.

fixes #585839
 2009-08-11 02:30:45 +01:00
Tim-Philipp Müller
c5793a6a45 Make build without warnings with debugging disabled 2009-08-11 02:30:45 +01:00
Olivier Crête
cf873498d2 rtpbin: Transform the right session sdes message 
Fixes #584165
 2009-08-11 02:30:45 +01:00
Olivier Crête
dee142a945 Add ssrc to application/x-rtp-source-sdes structure 2009-08-11 02:30:45 +01:00
Wim Taymans
bf15048f42 rtpsouce: the network address is in network order 
Bring the network address in netowkr byte order to the host order.
 2009-08-11 02:30:45 +01:00
Wim Taymans
91eef69131 rtpsource: byteswap the port from GstNetAddress 
Since the port in GstNetAddress is in network order we might need to byteswap it
before adding it to the source statistics.
 2009-08-11 02:30:45 +01:00
Wim Taymans
51251d0fa8 rtpbin: remove ptdemux ghostpads 2009-08-11 02:30:44 +01:00
Wim Taymans
7d9c2d20df rtpbin: add to new signal to remove SSRC pads 2009-08-11 02:30:44 +01:00
Ali Sabil
6c684e59c6 ssrcdemux: emit signal when pads are removed 
Add action signal to clear an SSRC in the ssrc demuxer.
Add signal to notify of removed ssrc.

See #554839
 2009-08-11 02:30:44 +01:00
Wim Taymans
48872d8215 rtpbin: use our ghostpads instead of its target 
Since we keep a reference to our ghostpads, we can use them to track sessions.
This avoid us having to mess with the target of the ghostpad.
 2009-08-11 02:30:44 +01:00
Wim Taymans
901b7f3b69 rtpbin: don't warn when getting request pads twice 
Allow getting the request pads multiple times, just return the previously
created pads.
 2009-08-11 02:30:44 +01:00
Wim Taymans
0ae6e3603b rtpsource: add RTP and RTCP source address 
Add the RTP and RTCP sender addresses in the stats structure.
 2009-08-11 02:30:44 +01:00
Wim Taymans
62727e8fab rtpsession: reuse source code for SDES 
Reuse the RTPSource object property instead of duplicating code.
 2009-08-11 02:30:44 +01:00
Wim Taymans
1719af9113 rtpbin: set target state on new elements 
Set the state on newly added elements to the state of the parent.
Add some debug info and do some cleanups
 2009-08-11 02:30:43 +01:00