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13451 commits 44 branches 1508 tags 536 MiB
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Author SHA1 Message Date
Tim-Philipp Müller
c3af414cbf multipartdemux: remove dynamic sourcpads when going from PAUSED to READY 2013-08-23 15:57:46 +01:00
Tim-Philipp Müller
7d78a68c8d multipartdemux: timestamp output buffers based on first input buffer that provided bytes not last 
https://bugzilla.gnome.org/show_bug.cgi?id=637754
 2013-08-23 15:57:46 +01:00
Wim Taymans
54e7e7547a rtxqueue: add property to configure queue size 2013-08-23 15:47:25 +02:00
Wim Taymans
43359b9244 tests: add retransmission example 2013-08-23 12:10:19 +02:00
Wim Taymans
84833bed11 rtpbin: proxy jitterbuffer do-retransmission property 2013-08-23 12:10:19 +02:00
Michael Olbrich
23d4044e2c avimux: unmap the correct buffer 
The audio buffer was mapped so unmap it and not the video buffer

https://bugzilla.gnome.org/show_bug.cgi?id=706642
 2013-08-23 11:32:52 +02:00
Olivier Crête
e00b8f0a4a pulsesink: Add property to find out the device currently in use 
https://bugzilla.gnome.org/show_bug.cgi?id=590768
 2013-08-22 15:03:29 -04:00
Olivier Crête
d379e237c1 pulsesink: De-duplicate code to get the current sink input info 
https://bugzilla.gnome.org/show_bug.cgi?id=590768
 2013-08-22 14:37:28 -04:00
Olivier Crête
8f9fbfa992 pulsesink: Implement changing the device while playing 
https://bugzilla.gnome.org/show_bug.cgi?id=590768
 2013-08-22 14:25:30 -04:00
Olivier Crête
691b04e5c9 pulsesrc: Add property to find out the device currently in use 
https://bugzilla.gnome.org/show_bug.cgi?id=590768
 2013-08-22 13:32:04 -04:00
Olivier Crête
d56f4718c2 pulsesrc: De-duplicate code to get the current source output info 
https://bugzilla.gnome.org/show_bug.cgi?id=590768
 2013-08-22 13:32:04 -04:00
Olivier Crête
c3642e3ecf pulsesrc: Implement changing the device while playing 
https://bugzilla.gnome.org/show_bug.cgi?id=590768
 2013-08-22 13:32:04 -04:00
Sebastian Dröge
e98767e864 configure: Fix bz2 configure check for Windows 
Due to function decorations on Windows AC_CHECK_LIB can't be used to check for bz2.

https://bugzilla.gnome.org/show_bug.cgi?id=465924
 2013-08-22 14:55:14 +02:00
Akihiro Tsukada
fda72021d2 pulsesink: Add support for AAC pass-through 
https://bugzilla.gnome.org/show_bug.cgi?id=694445
 2013-08-21 21:48:56 +02:00
Kishore Arepalli
1e917822a9 gdkpixbufoverlay: crashes if any property changes during playback when location property is not set 
https://bugzilla.gnome.org/show_bug.cgi?id=702988
 2013-08-21 21:42:02 +02:00
Olivier Crête
db84b928a3 pulse: Share static caps definition between src and sink 
The src was also missing 24-bit sample formats
 2013-08-21 15:23:12 -04:00
Wim Taymans
89b9019e3e rtx: various improvements 
Use locking
Don't push from the event handler, collected packets in a queue and push from
the chain function.
Clear queues on shutdown.
 2013-08-21 17:02:27 +02:00
Wim Taymans
ee15bc9284 session: generate events correctly 
Do correct shifting of the bitmask for lost packets.
 2013-08-21 17:02:27 +02:00
Wim Taymans
67523d3ecb rtp: register rtx element better 2013-08-21 17:02:26 +02:00
Sebastian Dröge
7692e9e569 directsoundsink: WAVEFORMATEX is unsigned for 8 bit integers, and signed for others 
Probably fixes
https://bugzilla.gnome.org/show_bug.cgi?id=705477
 2013-08-21 16:41:21 +02:00
Tim-Philipp Müller
cebfacd1fa jpegenc: don't ignore return value from _finish_frame() 
gst_video_encoder_finish_frame() will return FLOW_OK here if
there's no output buffer.
 2013-08-21 13:05:00 +01:00
Wim Taymans
f626e29897 jpegdepay: add some more debug 2013-08-21 12:56:35 +02:00
Wim Taymans
77ed44a88a rtpgstdepay: only push events when they changed 
Keep track of the STREAM_START and TAG events and only push them
when they changed.
 2013-08-21 12:10:00 +02:00
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