Vincent Penquerc'h
c0e101e93f
various: fix pad template leaks
...
https://bugzilla.gnome.org/show_bug.cgi?id=662664
2011-11-28 13:30:27 +00:00
Tim-Philipp Müller
09ca5fa910
rtpmanager: rename gstrtp* -> rtp*
...
This was done in 0.10 to avoid conflict with the rtp elements in
farsight, but the gst-prefixing is no longer needed in 0.11
2011-11-24 00:54:08 +00:00
Matej Knopp
1e5dd9e315
Fix printf format compiler warnings on OS X / 64bit
...
https://bugzilla.gnome.org/show_bug.cgi?id=662615
2011-11-22 01:28:22 +00:00
Wim Taymans
f8e988a94c
update for activation changes
2011-11-21 13:37:01 +01:00
Wim Taymans
b7aa7bca52
add parent to activate functions
2011-11-18 13:57:20 +01:00
Wim Taymans
07cc855b24
Merge branch 'master' into 0.11
...
Conflicts:
ext/speex/gstspeexenc.c
gst/rtpmanager/rtpsession.c
2011-11-17 17:17:11 +01:00
Wim Taymans
105650127e
add parent to pad functions
2011-11-17 15:02:55 +01:00
Wim Taymans
7cc4b72550
add parent to internal links
2011-11-16 17:54:49 +01:00
Wim Taymans
6190312214
add parent to query function
2011-11-16 17:27:13 +01:00
Wim Taymans
797523efbd
_peer_get_caps() -> _peer_query_caps()
2011-11-15 18:04:44 +01:00
Wim Taymans
75dc9634eb
change getcaps to query
...
Chain up event function in payloaders.
2011-11-15 18:04:44 +01:00
Olivier Crête
1169bb05af
gstrtpsession: Add special mode to use FIR as repair as Google does
...
https://bugzilla.gnome.org/show_bug.cgi?id=658419
2011-11-14 12:26:27 +01:00
Olivier Crête
79a9564c68
rtpsession: Send FIR requests in response to key unit requests with all-headers=TRUE
...
https://bugzilla.gnome.org/show_bug.cgi?id=658419
2011-11-14 12:26:27 +01:00
Olivier Crête
12a6b9613b
rtpsession: Put the PLI requests in each RTPSource
...
Also refactor a bit and put all the keyframe request code in one
place inside rtpsession.c
https://bugzilla.gnome.org/show_bug.cgi?id=658419
2011-11-14 12:26:27 +01:00
Olivier Crête
59c028a4ce
rtpsession: Hack to FIR because Google doesn't set the sender ssrc correctly
...
https://bugzilla.gnome.org/show_bug.cgi?id=658419
2011-11-14 12:26:27 +01:00
Olivier Crête
0ad78db0a3
rtpsession: Process received Full Intra Requests
...
Process FIR requests according to RFC 5104
https://bugzilla.gnome.org/show_bug.cgi?id=658419
2011-11-14 12:26:27 +01:00
Wim Taymans
a19a4a69ae
more template fixes
2011-11-04 13:12:37 +01:00
Wim Taymans
a95acb7122
make %u in all request pad templates
2011-11-04 11:58:22 +01:00
Wim Taymans
6cbd6afc0b
update for new net library
2011-11-03 16:43:00 +01:00
Wim Taymans
83ccefb24e
update for netbuffer api change
2011-11-02 09:06:38 +01:00
Wim Taymans
75e0c6052f
update for netaddress change
2011-11-02 09:06:38 +01:00
Wim Taymans
9a8a8e72c8
structure: fix for api update
2011-11-02 09:06:37 +01:00
Wim Taymans
161310fa23
bufferlist: update for new API
2011-11-02 09:06:37 +01:00
Tim-Philipp Müller
d18a578ba4
rtpmanager, v4l2: fix compiler warnings after gst_caps_new_simple() change
2011-10-28 09:06:41 +01:00
Wim Taymans
fc4684f4c6
fix compilation
2011-10-27 16:03:17 +02:00
Edward Hervey
d4a2a46606
rtpssrcdemux: Fix wrong usage of gst_iterator_filter
...
It takes a GValue* as the user_data.
And don't forget to unref the demuxer before returning.
2011-10-13 09:34:04 +02:00
Wim Taymans
87fbd1e784
Merge branch 'master' into 0.11
...
Conflicts:
common
ext/pulse/pulsesink.c
ext/soup/gstsouphttpclientsink.c
gst/audioparsers/gstaacparse.c
gst/audioparsers/gstac3parse.c
gst/rtp/gstrtph264depay.c
gst/rtpmanager/gstrtpjitterbuffer.c
gst/rtpmanager/rtpjitterbuffer.c
gst/rtsp/gstrtspsrc.c
sys/ximage/gstximagesrc.c
2011-09-28 12:44:59 +02:00
Ha Nguyen
931020158e
rtpbin: Fix a leaked clock for each buffering message
...
Fixes bug #659237 .
2011-09-19 14:05:26 +02:00
Mark Nauwelaerts
e2179cbb74
rtpsession: avoid source premature timing out
...
Use slightly adjusted sender interval to determine sender timeout rather than
our own sender side interval (which may have been forced small).
2011-09-19 11:56:44 +02:00
Mark Nauwelaerts
f65d4c8300
rtpsession: avoid timing out source too quickly
...
... following a PAUSE/PLAY cycle, particularly applicable when operating
with a short RTCP interval (possibly forced so server-side).
2011-09-19 11:56:44 +02:00
Mark Nauwelaerts
77ebd33991
rtpjitterbuffer/rtpbin: relax dropping rtcp packets
...
... to at least having it trigger a/v synchronization, possibly without
using provided values which are still not considered sane
(as previously dropped).
2011-09-19 11:56:44 +02:00
Mark Nauwelaerts
adfe7d0467
rtpjitterbuffer: some more reset when clearing pt map
...
... which in particular caters for some more reset following a possible
rtsp PLAY.
2011-09-19 11:56:44 +02:00
Mark Nauwelaerts
915db26029
rtpjitterbuffer: only reset skew on gap if input ts available
2011-09-19 11:56:44 +02:00
Mark Nauwelaerts
1e17e10f75
rtpjitterbuffer: check some more for possible rtp timestamp discontinuity
...
... when operating in non slave mode, and reset if detected.
This should avoid some (large) bogus outgoing timestamp due to jumps
in rtp time, as result of PAUSE/PLAY or seek or ...
2011-09-19 11:56:40 +02:00
Mark Nauwelaerts
9c95072048
rtpbin: alternative inter-stream syncing methods
...
... at least if not syncing to NPT time:
* either sync using RTCP SR data (as currently)
* only perform the above once using initial RTCP SR packets
* discard RTCP and sync by equating provided stream's clock-base rtptime,
as provided by jitterbuffer (typically obtained from RTP-Info in RTSP).
2011-09-19 11:52:03 +02:00
Mark Nauwelaerts
4b7301e4d1
rtpjitterbuffer: also provide clock-base to sync signal
2011-09-19 11:52:00 +02:00
Mark Nauwelaerts
f29c253934
rtpbin: allow configurable rtcp stream syncing interval
...
... rather than necessarily syncing at each RTCP SR.
2011-09-19 11:51:57 +02:00
Mark Nauwelaerts
afd26f0078
rtpsession: trigger reconsideration if rtcp interval set
2011-09-19 11:51:50 +02:00
Wim Taymans
33f18b8ea4
Merge branch 'master' into 0.11
...
Conflicts:
gst/audioparsers/gstamrparse.c
gst/isomp4/qtdemux.c
2011-09-06 16:06:25 +02:00
Olivier Crête
b2e8362767
rtpsession: Initialise the last_keyframe_request variable
2011-09-02 19:24:46 -04:00
Wim Taymans
4121021bb2
Merge branch 'master' into 0.11
...
Conflicts:
ext/pulse/pulsesink.c
ext/pulse/pulsesrc.c
gst/audioparsers/gstac3parse.c
gst/rtp/gstrtph264depay.c
gst/rtp/gstrtph264pay.c
gst/rtpmanager/gstrtpssrcdemux.c
2011-08-03 18:25:30 +02:00
Mark Nauwelaerts
c03648c8bb
rtpsession: properly init rtcp_min_interval
2011-07-29 12:08:42 +02:00
Mark Nauwelaerts
3a98f6f0fd
rtpssrcdemux: keep a ref on the src pad while using it
...
Prevent a possible race if clear_ssrc() is called between getting the pad and
doing the push.
Based on patch by <olivier.crete@collabora.com>
https://bugzilla.gnome.org/show_bug.cgi?id=650916
2011-07-28 14:51:01 +02:00
Olivier Crête
c7b9b98648
rtpssrcdemux: Make the pads lock recursive and hold it across the signal emit
...
We need to keep the lock held because we don't want a push before the "new-ssrc-pad"
handler has completed. But we may want to push an event from inside that handler, hence
the recursive mutex.
https://bugzilla.gnome.org/show_bug.cgi?id=650916
2011-07-28 14:50:59 +02:00
Olivier Crête
e26b5391c2
rtpssrcdemux: Use PADs lock
...
https://bugzilla.gnome.org/show_bug.cgi?id=650916
2011-07-28 14:50:57 +02:00
Olivier Crête
6095d2a3f0
rtpsession: Always send application requested feedback in immediate mode
...
Send as many application requested feedback messages in immediate mode, even if they
have already been sent.
https://bugzilla.gnome.org/show_bug.cgi?id=654583
2011-07-25 17:20:59 +02:00
Olivier Crête
354faabda0
rtpsession: Don't let the computed RTP bandwidth fall too low
...
If it falls too low, the computed RTCP bandwidth will be near zero and
the RTCP thread will be stopped.
https://bugzilla.gnome.org/show_bug.cgi?id=654583
2011-07-25 16:19:00 +02:00
Olivier Crête
4d48109f9d
rtpsession: Wait longer to timeout SSRC collision
...
Using the current RTCP interval to timeout SSRC collision can lead to
collisions being timed out immediately if a BYE packet is sent because
it is sent immediately, so the interval is 0. This is not what we
want. So just set a static 10 times the default RTCP interval, it
should be enough
https://bugzilla.gnome.org/show_bug.cgi?id=648642
2011-07-25 16:18:58 +02:00
Mark Nauwelaerts
ef02634dc6
rtpmanager: port to 0.11
...
* use G_DEFINE_TYPE
* adjust to new GstBuffer and corresponding rtp and rtcp buffer interfaces
* misc caps and segment handling changes
FIXME: also relies on being able to pass caps along with a buffer,
which has no evident equivalent yet, so that either needs one,
or still needs quite some code path modification to drag along caps.
2011-07-06 10:16:12 +02:00
Mark Nauwelaerts
d59a00aa1c
Merge branch 'master' into 0.11
...
Conflicts:
ext/pulse/pulsesink.c
2011-07-04 11:48:13 +02:00
Miguel Angel Cabrera Moya
977a5eee7a
rtpjitterbuffer: return correct type when assertion fails
2011-06-24 11:59:01 +02:00
Wim Taymans
cc65bff7c1
Merge branch 'master' into 0.11
...
Conflicts:
configure.ac
docs/plugins/inspect/plugin-esdsink.xml
docs/plugins/inspect/plugin-gconfelements.xml
2011-06-21 18:24:41 +02:00
Olivier Crête
581a30d892
rtpsession: The signal has 5 arguments, not 4
2011-06-20 16:47:36 -04:00
Wim Taymans
409f29700d
-good: port some more plugins
2011-06-13 17:51:40 +02:00
Wim Taymans
e15651816e
Merge branch 'master' into 0.11
2011-05-17 16:13:59 +02:00
Sebastian Dröge
b694bfeca3
ssrcdemux: Fix uninitialized variable compiler warning for (pre-) releases too
2011-05-17 10:47:32 +02:00
Sebastian Dröge
0f05d3e5a5
rtpssrcdemux: Fix uninitialized variable compiler warning
2011-05-17 09:24:08 +02:00
Olivier Crête
b6bfc512e8
ssrcdemux: Implement iterate internal links for sink pads
...
https://bugzilla.gnome.org/show_bug.cgi?id=649617
2011-05-17 09:22:29 +02:00
Olivier Crête
23b6c8febc
rtpssrcdemux: iterate pad function is only valid for src pads
...
The iterate function is only used for src pads, so mark it as such and remove
dead code.
https://bugzilla.gnome.org/show_bug.cgi?id=649617
2011-05-17 09:22:25 +02:00
Olivier Crête
1bf94a92b0
rtpssrcdemux: Release lock before emitting signal
...
If the lock is not released before emitting a signal, it may cause a deadlock
if any other function in the element is called.
Also removed an unused timestamp parameter
https://bugzilla.gnome.org/show_bug.cgi?id=649617
2011-05-17 09:22:20 +02:00
Wim Taymans
a1894ed363
Merge branch 'master' into 0.11
2011-04-25 11:38:28 +02:00
Olivier Crête
42531337f5
rtpsession: Remove incomplete support for RTCP FIR
...
Remove bits that were meant to suppport RTCP FIR
https://bugzilla.gnome.org/show_bug.cgi?id=648160
2011-04-20 07:50:43 +01:00
Wim Taymans
7555d0949f
Merge branch 'master' into 0.11
...
Conflicts:
android/apetag.mk
android/avi.mk
android/flv.mk
android/icydemux.mk
android/id3demux.mk
android/qtdemux.mk
android/rtp.mk
android/rtpmanager.mk
android/rtsp.mk
android/soup.mk
android/udp.mk
android/wavenc.mk
android/wavparse.mk
configure.ac
2011-04-18 10:23:45 +02:00
Robert Swain
5b18c652fb
rtp, rtpmanager: Address unused but set variables
...
GCC 4.6.x spits warnings about variables that are unused but set. Such
variables have been removed where trivial but with comments left behind
for informational purposes in some cases.
gst_rtp_session_chain_recv_rtcp () was changed in commit 490113d4
to always return GST_FLOW_OK instead of the return value of
rtp_session_process_rtcp (), so we'll keep it that way.
2011-04-16 12:49:16 +01:00
Olivier Crête
9d9257916b
rtpsession: Use existing functions to parse RTCP FB packets
...
Use existing functions to get the FCI from FB packets.
https://bugzilla.gnome.org/show_bug.cgi?id=622553
2011-04-15 12:48:04 +01:00
Olivier Crête
5ccd964d86
rtpsession: marshal GstBuffer as a MiniObject instead of a pointer
...
https://bugzilla.gnome.org/show_bug.cgi?id=622553
2011-04-15 12:47:40 +01:00
Pascal Buhler
0d2d52856f
rtpssrcdemux: Unknown SSRC is not fatal
...
https://bugzilla.gnome.org/show_bug.cgi?id=646966
2011-04-11 17:37:58 -04:00
Pascal Buhler
58ef84846e
rtpsession: Number of active sources should be updated whenever the status of the source changes to active
...
Forward-ported by Olivier Crête
https://bugzilla.gnome.org/show_bug.cgi?id=646965
2011-04-11 17:37:36 -04:00
Havard Graff
53c88ae33e
rtpmanager: ignore a BYE if it is sent with our internal SSRC
...
https://bugzilla.gnome.org/show_bug.cgi?id=646964
2011-04-11 17:34:12 -04:00
Thibault Saunier
b541208b77
android: Make it ready for androgenizer
...
Remove the android/ top dir
Fixe the Makefile.am to be androgenized
To build gstreamer for android we are now using androgenizer which generates the needed Android.mk files.
Androgenizer can be found here: http://git.collabora.co.uk/?p=user/derek/androgenizer.git
2011-04-11 01:20:11 +02:00
Havard Graff
93f022d6ab
rtpsession: fix wrongly applied patch
...
Obviously recv_rtp_sink does not have much to do with send_rtcp_src...
See commit 046ff170.
https://bugzilla.gnome.org/show_bug.cgi?id=647263
2011-04-09 12:32:37 +01:00
Havard Graff
e71a908d96
jitterbuffer: Make src_query MT-safe
...
It is possible that the element might be going down while the event arrives
2011-04-08 15:23:05 +02:00
Sebastian Dröge
4c36ca30b2
jitterbuffer: Unref event if the parent element disappeared
2011-04-08 15:22:19 +02:00
Havard Graff
342686bb02
jitterbuffer: Make upstream events MT-safe
2011-04-08 15:21:46 +02:00
Sebastian Dröge
31af4fe33e
rtp: Unref events if the parent element disappeared
2011-04-08 15:20:51 +02:00
Ole André Vadla Ravnås
046f170d6a
rtpmanager: fix pad callbacks so they handle when parent goes away
...
1) We need to lock and get a strong ref to the parent, if still there.
2) If it has gone away, we need to handle that gracefully.
This is necessary in order to safely modify a running pipeline. Has been
observed when a streaming thread is doing a buffer_alloc() while an
application thread sends an event on a pad further downstream, and from
within a pad probe (holding STREAM_LOCK) carries out the pipeline plumbing
while the streaming thread has its buffer_alloc() in progress.
2011-04-08 15:16:56 +02:00
Havard Graff
f8370bb2a8
rtpsession: make iterate_internal_links MT-safe
2011-04-08 14:41:34 +02:00
Wim Taymans
c124ba1489
Merge branch 'master' into 0.11
...
Conflicts:
gst/rtsp/gstrtspsrc.c
2011-04-05 17:20:08 +02:00
Mark Nauwelaerts
e5bcaa45e6
Revert "jitterbuffer: reset element base_time upon flush"
...
This reverts commit f84b8a69cb
.
Fixes bug #646397 .
2011-04-04 11:49:00 +02:00
Wim Taymans
8f22a09dc4
Merge branch 'master' into 0.11-fdo
2011-03-28 20:50:59 +02:00
Mark Nauwelaerts
6bc1aa0e59
jitterbuffer: handle position query
2011-03-09 17:18:08 +01:00
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