Commit graph

408 commits

Author SHA1 Message Date
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