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
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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
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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
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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
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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