mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-26 02:00:33 +00:00
Code Issues 1 Projects Releases Wiki Activity
13578 commits 44 branches 1510 tags 328 MiB
Commit graph

13578 commits

This branch
This branch
All branches
Author SHA1 Message Date
Wim Taymans
b144809b7c rtpgstpay: taglists should not be merged in 1.0 2013-08-21 10:52:59 +02:00
Wim Taymans
69b0dcd7df rtpgstdepay: flush on FLUSH_STOP event 2013-08-21 10:28:50 +02:00
Wim Taymans
5ff9093843 rtpgstpay: reset on state change 
Do full reset on state change to READY
 2013-08-21 10:03:52 +02:00
Wim Taymans
ae9239aac7 rtpgstpay: reset on FLUSH_STOP 
Clear the adapter and pending buffer list on FLUSH_STOP.
 2013-08-21 09:55:20 +02:00
Wim Taymans
2e8955df39 rtpgstpay: don't use clock for config interval 
We can't use the clock to time our config-interval because we are not
live (or there might not be a clock or the clock might not be running).
Instead just simply take the timestamp diff.
 2013-08-21 09:39:30 +02:00
Wim Taymans
182f96ff79 rtpgstay: don't use // comments 2013-08-21 09:33:04 +02:00
Youness Alaoui
e22f7e91c4 rtspsrc: Fix response argument in handle-request signal 2013-08-21 09:06:02 +02:00
Youness Alaoui
6636efd31a rtspsrc: Add sdes property and proxy it to rtpbin 2013-08-21 09:06:02 +02:00
Youness Alaoui
62a6f58697 Send a stream-start whenever we send tags 
This is to make sure tags are cleared on the client if the
stream-start was previously lost, otherwise, the client may end
up with a merged taglist of multiple songs
 2013-08-21 09:06:01 +02:00
Youness Alaoui
05bcfee5a3 rtpgstpay: Add a config-interval property to resend the caps/tags at a regular interval 
This is useful in case the packet containing the inlined caps was lost
or if new client joins an already running RTP stream and they missed
the previous tag events.
This also makes the payloader keep a list of merged tags so the retransmitted
tag event contains all previously received. A STREAM_START event will
flush the list of tags.
 2013-08-21 09:06:01 +02:00
Youness Alaoui
1f4ca28868 rtpgstpay: Refactor the setcaps and use new method to send arbitrary caps at any time 2013-08-21 09:06:01 +02:00
Youness Alaoui
9257409613 rtpgstpay: Do not flush events for stream-start and avoid conflict between event and pending inline caps 2013-08-21 09:06:01 +02:00
Youness Alaoui
2d53289b6b rtpgstpay: Add a create_from_adapter API and use a list of GstBufferList 
This is necessary to fix event/caps sending. If we send a STREAM_START
packet, it will cause an error because the stream didn't receive its
caps and new-segment events, so we must wait for the first buffer before
sending the stream-start event buffer. However, the caps will be sent
at the same time and so the 'inline caps' will be set for the event.
We need to be able to payload individual packets (data, caps or events)
and only send them when we call flush.
 2013-08-21 09:06:01 +02:00
Youness Alaoui
0070ba76f2 rtpgstpay: Add etype=4 for payloading GST_EVENT_STREAM_START 2013-08-21 09:06:01 +02:00
Youness Alaoui
6155b27971 rtpgstpay: Fix typo, GST_EVENT_CUSTOM_BOTH has etype of 3 2013-08-21 09:06:01 +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
Olivier Crête
d595c08aca v4l2: Add forward declaration for gst_v4l2_object_get_format_list 2013-08-19 13:19:42 -04:00
Olivier Crête
48caa1712a v4l2: De-duplicate caps probing between src and sink 2013-08-19 13:08:18 -04:00
Olivier Crête
dd5d93f0f6 pulse: Remove unused GstPulseProbe 2013-08-19 12:56:27 -04:00
Olivier Crête
24286f1612 v4l2: Use G_DEFINE_ macros for added thread safety 2013-08-19 12:48:35 -04:00
Thibault Saunier
e47ffb203b videomixer: Do not send flush_stop ourself after a flush_start 
When we receive a flush_start, we should wait for the next flush_stop
and foward it, not create a flush_stop ourself.
 2013-08-17 11:40:27 +02:00
Wim Taymans
db90f6e68d h264depay: init debug category early 
Init the debug variable when we register the element because it is also used by
the payloader element when it calls the add_sps_pps method.
 2013-08-16 17:12:19 +02:00
Sebastian Dröge
de7e1cb6dd flacenc: Properly set headers via the base class instead of just pushing them downstream 
Prevents buffers from being send before the caps and segment events.
 2013-08-16 13:26:50 +02:00
Chris Bass
3e9dea3f8c qtdemux: check denominator isn't zero before scaling duration. 
When gst_qtdemux_configure_stream sets fps_d, check that n_samples is
non-zero before using it as a denominator to scale the stream duration.

https://bugzilla.gnome.org/show_bug.cgi?id=706076
 2013-08-16 10:14:30 +02:00
Sebastian Dröge
b1e442236f ext: Use new flush vfunc of video codec base classes and remove reset implementations 2013-08-15 15:08:05 +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
Tim-Philipp Müller
1095114d89 configure: require libsoup >= 2.38 
Bump libsoup requirement for newer API used, like headers_get_one().
2.38 is from early 2012 and is in linen with our GLib requirement.
 2013-08-14 13:10:32 +01:00
Tim-Philipp Müller
604bfa586e soup: don't use deprecated soup_message_headers_get() API 2013-08-14 11:54:19 +01:00
Edward Hervey
f996daf622 .gitignore: Ignore files from automake test-driver 2013-08-13 17:44:50 +02:00
Olivier Crête
4c6e636720 rtph264pay: Use the SPS/PPS handling function from the depayloader 
Remove duplicated copies

https://bugzilla.gnome.org/show_bug.cgi?id=705553
 2013-08-13 10:38:23 -04:00
Olivier Crête
742b90747d rtph264depay: Make the SPS/PPS deduplication function generic 
Make it not touch any internals of the depayloader

https://bugzilla.gnome.org/show_bug.cgi?id=705553
 2013-08-13 10:38:23 -04:00
Chris Bass
b40bf67526 aacparse: allow conversion from raw AAC to ADTS 
This patch will prepend ADTS headers to raw AAC audio frames, allowing
upstream elements to link to decoders that only support AAC in ADTS format.

Note that no error correction bits are added to ADTS frames in this code.

https://bugzilla.gnome.org/show_bug.cgi?id=615740
 2013-08-13 15:58:23 +02:00
Sebastian Dröge
282afae244 rtspsrc: Only free GCheckSum after its last usage 
https://bugzilla.gnome.org/show_bug.cgi?id=705760
 2013-08-13 12:44:11 +02:00
Andoni Morales Alastruey
24b4ea3418 souphttpsrc: fix critical setting a NULL uri redirection 2013-08-13 12:02:29 +02:00
Andoni Morales Alastruey
b9faeab236 souphttpsrc: add redirection to the URI query 2013-08-12 16:40:26 +02:00