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Author SHA1 Message Date
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
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
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
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
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
Matej Knopp
2269ac8f28 qtdemux: elst should offset samples instead of buffers 
The current approach where buffers are offset is not ideal, as during seek
and loop current time is compared to sample times.

https://bugzilla.gnome.org/show_bug.cgi?id=700264
 2013-08-12 13:48:04 +02:00
Thibault Saunier
6c349d6ec3 videomixer: Send EOS if buf_end >= segment.stop 
That means the whole segment is already played, and we are sure we
are EOS at that point.

Also handle segment seeks, and do not send EOS in that case.
 2013-08-11 19:05:18 +02:00
Matej Knopp
96afba915a avidemux: send proper stream_start event 
https://bugzilla.gnome.org//show_bug.cgi?id=705449
 2013-08-08 11:57:32 +02:00
Sebastian Dröge
9863e08839 matroskademux: Don't print warnings during flushing and stop as soon as possible 
https://bugzilla.gnome.org//show_bug.cgi?id=705442
 2013-08-08 11:53:15 +02:00
Tim-Philipp Müller
957c8e3e61 rtpvp8depay: mark key frames and delta frames properly 
https://bugzilla.gnome.org/show_bug.cgi?id=705550
 2013-08-07 11:14:38 +01:00
Wim Taymans
48174164eb session: add NACK feedback in RTCP 2013-08-06 15:50:19 +02:00
Wim Taymans
4379ed1dee source: add methods to register NACK 
Add a method to register a missing packet for an ssrc along with
methods to get the missing packets and clear them.
 2013-08-06 15:50:19 +02:00
Wim Taymans
50638b8106 session: handle Retransmission event and schedule NACK 
Handle the retransmission event from downstream and use it to schedule a NACK
request.
 2013-08-06 15:50:19 +02:00
Wim Taymans
0bddbd682d session: pass data to remove func 
Pass the data to the remove function because we are going to deref it when there
is pli or fir.
 2013-08-06 15:50:19 +02:00
Thibault Saunier
38946bd9f4 qtdemux: Fix compilation 2013-08-06 15:31:38 +02:00
Thibault Saunier
593a31f2b4 qtdemux: Raw buffer DTS should always be CLOCK_TIME_NONE 2013-08-06 15:17:44 +02:00
Thibault Saunier
c5fa4666b7 videomixer: Make sure to send EOS if the buffer end time equals the segment end time 
Otherwize EOS never gets sent in that particular case.
 2013-08-06 12:21:33 +02:00
Sjoerd Simons
d14d4c436c goom: Ensure src caps are writable 
In some cases the src caps determined by goom weren't writable, causing
a bunch of assertion failures and failed caps. Fixed by always
explicitely making the caps writable

https://bugzilla.gnome.org/show_bug.cgi?id=705475
 2013-08-05 15:33:39 +02:00
Wim Taymans
3c82de59f9 session: use common send_rtcp method 
Reuse the send_rtcp method that already asks for the current time when
requesting a keyframe.
 2013-08-05 15:02:59 +02:00
Wim Taymans
3c14c6021c session: Don't use ClockTimeDiff for unsigned delays 2013-08-05 15:02:59 +02:00
Edward Hervey
4f4f6432cc qtmux: Use buffer PTS if DTS is not set 
Avoids ending up with completely bogus scaled duration/pts when new
buffers have invalid DTS.
 2013-08-04 17:15:38 +02:00
Tim-Philipp Müller
7272dec5fe rtpdec: use generic marshaller 2013-08-04 11:20:41 +01:00
Tim-Philipp Müller
fe098e3aff udp: remove unused marshal and enumtypes files 2013-08-04 11:03:07 +01:00
Tim-Philipp Müller
7469cd3a4c rtpmanager: use generic marshaller 2013-08-04 11:03:07 +01:00
Wim Taymans
7584f91f31 jitterbuffer: send event in right direction 2013-08-04 00:24:36 +02:00
Wim Taymans
9613e481ad session: add FIR and PLI like other RTCP packets 
Add the FIR and PLI packets like the other RTCP packet instead of from the
on-sending-rtcp default signal handler.
 2013-08-03 00:33:24 +02:00
Wim Taymans
743e1b1191 jitterbuffer: fix property ranges 2013-08-02 17:22:55 +02:00
Wim Taymans
cd0164f4cc jitterbuffer: push retransmission events 2013-08-02 16:43:59 +02:00
Wim Taymans
9a13267e85 jitterbuffer: add support for retransmission retry 
When we didn't receive a packet after requesting retransmission, retry
asking for retransmission for a certain period.
 2013-08-02 14:54:56 +02:00
Wim Taymans
e9ad5126db jitterbuffer: add properties 
Add properties to control retransmission parameters
 2013-08-02 14:47:56 +02:00
Wim Taymans
a8c7ff7489 jitterbuffer: use corrected timeout when rescheduling 
When we recalculate the timeout, use the corrected timeout value depending on
the timer type.
 2013-08-02 12:44:58 +02:00
Wim Taymans
9c7e3e3455 jitterbuffer: update timers after queueing 
Else we might update the timer needlessly for duplicates.
 2013-08-02 12:43:00 +02:00
Wim Taymans
ebd6b8f8ab jitterbuffer: move method up 2013-08-02 12:42:08 +02:00
Wim Taymans
f6b6797874 jitterbuffer: small cleanup 2013-08-02 06:28:32 +02:00
Wim Taymans
0e41414926 jitterbuffer: unschedule old expected packets 
When we receive a new packet, unschedule old outstanding packets when their
seqnum is too far away.
 2013-08-01 23:36:07 +02:00
Wim Taymans
70695466ed jitterbuffer: refactor timer update 2013-08-01 23:32:00 +02:00
Wim Taymans
4ab3f5d3da jitterbuffer: update timers when removing 
Update the timers when we remove a timer.
Handle canceled timers, make them unschedule the current timer and
trigger the timeout code.
 2013-08-01 23:24:29 +02:00
Wim Taymans
b983cf675b jitterbuffer: fix typo 2013-08-01 23:22:02 +02:00
Wim Taymans
f3c658cbe6 jitterbuffer: improve timeout management 
If we change the seqnum of an existing timer and we were waiting for
that timer, unschedule it. If we change the timeout of an existing timer and we
were waiting on it, only unschedule when the new time is smaller.
 2013-08-01 15:40:52 +02:00
Wim Taymans
77e5d320ab jitterbuffer: install timer for expected arrival 
Install a timer that is triggered when the expected arrival time of a packet
expired.
 2013-08-01 15:11:13 +02:00
Wim Taymans
f08d98404e jitterbuffer: improve unschedule of timers 
Conflicts:
	gst/rtpmanager/gstrtpjitterbuffer.c
 2013-08-01 14:57:11 +02:00
Wim Taymans
9d3b824e2a jitterbuffer: move code around 2013-08-01 12:21:53 +02:00
Wim Taymans
fe32e80c92 jitterbuffer: estimate inter packet spacing 
When we see two packets with consecutive seqnums and a different RTP time, use
the DTS difference as the inter packet spacing estimate.
 2013-08-01 12:07:11 +02:00
Wim Taymans
255b7106f5 jitterbuffer: keep track of current timeout 2013-08-01 12:01:15 +02:00
Wim Taymans
7e43dba19b jitterbuffer: cleanup timer handling 2013-08-01 11:49:10 +02:00
Wim Taymans
9d88ac9cbb jitterbuffer: reset is only possible with a GAP 2013-08-01 11:40:41 +02:00
Wim Taymans
f864131227 jitterbuffer: operate on DTS 
Make the jitterbuffer schedule the timeouts based on the DTS instead
of the PTS. This makes it all smoother with reordered frames and gives
the decoder time to reorder the frames in time.
 2013-08-01 11:36:56 +02:00
Wim Taymans
80c5934290 jitterbuffer: rename timout variable 2013-08-01 11:14:12 +02:00
Wim Taymans
aa951433ee jitterbuffer: small cleanup 2013-07-31 17:08:58 +02:00
Wim Taymans
69c78f72d5 jitterbuffer: block output in paused or buffering 2013-07-31 16:59:58 +02:00
Wim Taymans
4fbbc53a49 jitterbuffer: store pts in timer 
Only store the pts in the timer so that we can both do timeouts with timings on
the input and output of the jitterbuffer.
 2013-07-31 16:59:09 +02:00
Wim Taymans
77846d35c6 rtpjitterbuffer: refactor jitterbuffer 
Refactor the jitterbuffer code. Make separate function for peeking a buffer,
pushing the next buffer, waiting for timeouts and handling the timeouts.

The main loop now tries to push as many buffers as it can until it runs out of
buffers or when it detects a seqnum discont. Then it will wait for some event to
happen before attempting to push more buffers.

Make methods to register timeouts in an array. These timeouts are registered
when we detect a missing packet, sync for the first packet or when we find an
estimation for the end-of-stream.

This greatly simplifies and clarifies the code and also makes it possible to
register more complicated timeout schemes later.
 2013-07-30 23:24:23 +02:00
Wim Taymans
ea931d4f57 rtpjitterbuffer: use NULL to ignore percent 
If we pass NULL to pop and push we ignore the percent result.
 2013-07-30 23:24:23 +02:00
Wim Taymans
b3e8a85a54 jitterbuffer: refactor 
Move eos estimation into separate function
 2013-07-30 23:24:22 +02:00
Tim-Philipp Müller
a5532b4510 flvdemux: don't leak stream_id string 
https://bugzilla.gnome.org/show_bug.cgi?id=705142
 2013-07-30 14:28:19 +01:00
Sebastian Dröge
2e35b36aab gst: Don't swap start/stop for negative rates in the SEGMENT query 2013-07-29 12:12:41 +02:00
Matej Knopp
47ed79fb1c qtdemux: Check for data size when parsing h264 codec data from strf atom 2013-07-29 11:53:07 +02:00
Sebastian Dröge
722ef42196 matroskademux: Implement SEGMENT query 2013-07-29 10:53:54 +02:00
Sebastian Dröge
d135373beb flvdemux: Implement SEGMENT query 2013-07-29 10:53:47 +02:00
Sebastian Dröge
4e78974c87 avidemux: Implement SEGMENT query 2013-07-29 10:50:59 +02:00
Matej Knopp
2dcdfe07f7 qtdemux: Support H264 fourcc 
https://bugzilla.gnome.org/show_bug.cgi?id=704996
 2013-07-29 09:11:39 +02:00
Sebastian Dröge
1fbb6d30a6 avidemux: Fix duration reporting in push mode 
https://bugzilla.gnome.org/show_bug.cgi?id=700933
 2013-07-28 17:38:56 +02:00
Sebastian Dröge
89a3dc2ecd avidemux: Don't forget unmapping and unreffing buffer 2013-07-28 17:32:59 +02:00
Matej Knopp
1947587784 avidemux: unmap buffer 
https://bugzilla.gnome.org/show_bug.cgi?id=704951
 2013-07-28 17:32:59 +02:00
Wim Taymans
02359f9219 session: don't make buffer writable prematurely 
There is no reason to make the SR buffer writable at this point. This is better
delayed until needed.
 2013-07-26 22:31:41 +02:00
Wim Taymans
0261199fc4 session: ignore RTCP for inactive sources 2013-07-26 22:31:23 +02:00
Wim Taymans
a4b4ca53c0 session: small cleanup 2013-07-26 22:25:17 +02:00
Wim Taymans
e0abd2e9b5 session: handle partial RTCP report blocks 
When we have more SSRCs to report than what fit in an RTCP packet, use a
generation counter to make sure all of them end up in a packet eventually.
 2013-07-26 17:29:10 +02:00
Wim Taymans
6cce6fb04c session: create SSRC before doing session cleanup 
Make the internal source before we do session cleanup
 2013-07-26 17:29:10 +02:00
Wim Taymans
5b0298c63e session: reorganize the report block code 2013-07-26 17:29:10 +02:00
Matej Knopp
7335b81c47 matroskademux: fix memory leak in check_subtitle_buffer 
https://bugzilla.gnome.org/show_bug.cgi?id=704921
 2013-07-26 17:11:31 +02:00
Wim Taymans
3c44cd7c83 session: refactor active and sender checks 2013-07-26 14:21:40 +02:00
Wim Taymans
e952f7ba43 session: remove internal sources on timeout 
When an internal source times out and becomes a receiver, remove it.
 2013-07-26 12:18:01 +02:00
Wim Taymans
e9e2fe3950 session: create an internal source for RTCP 
When we need to do RTCP and we don't have an internal source yet,
make one.
 2013-07-26 12:18:01 +02:00
Wim Taymans
bd0709c15c session: remove old code to change SSRC 
Remove code used to change the SSRC after a collision. We now send
a RECONFIGURE event upstream to make the upstream element change the SSRC.
 2013-07-26 12:18:01 +02:00
Wim Taymans
88f5a5f355 source: don't update packet SSRC 
Remove the code to update the SSRC in packets, it can never be called now that
we always use a source with matching packet SSRC.
 2013-07-26 12:18:01 +02:00
Wim Taymans
abc90da1dc session: delay allocation of internal source 
Allocate the internal source when we receive a caps with the SSRC or when we see
a buffer with the SSRC.
 2013-07-26 12:18:01 +02:00
Wim Taymans
e0a1ce1291 session: generate reconfigure on collision 
When we detect a collision, change the SSRC that we suggest upstream
and trigger RECONFIGURE. This should make upstream select a new SSRC.
 2013-07-26 12:18:01 +02:00
Wim Taymans
495d43c089 session: produce RTCP for all internal sources 
Loop over all the internal sources and produce RTCP. We also need
to queue the RTCP packets and send them when we are finished.
 2013-07-26 12:18:00 +02:00
Wim Taymans
9505fd4150 session: deprecate internal source and ssrc properties 
Deprecate the internal source and internal ssrc properties. There might
be more than one internal source.
 2013-07-26 12:17:59 +02:00
Wim Taymans
3d6ee1fb5e session: internal sources don't use probation 2013-07-26 12:17:59 +02:00
Wim Taymans
0e53e9109e session: give caps to session 
Let the session parse the caps and update its SSRC when needed.
 2013-07-26 12:17:59 +02:00
Wim Taymans
c06482a2cb session: make method to suggest available SSRC 
Make a method to suggest the best available SSRC. This is the SSRC of the last
created internal source and is used to instruct upstream to produce this
SSRC.
 2013-07-26 12:17:59 +02:00
Wim Taymans
33ce50e8b1 session: keep SDES and set on new internal sources 
Keep track of the SDES ourselves and set it on all newly created
internal sources.
 2013-07-26 12:17:59 +02:00
Wim Taymans
5652f02b76 session: make method to make internal sources 
Add a method to obtain an internal source and use it to create
our internal source
 2013-07-26 12:17:59 +02:00
Wim Taymans
7f83927c95 session: count internal sources and how many are senders 2013-07-26 12:17:58 +02:00
Wim Taymans
719343c206 rtpsession: separate BYE marking and scheduling 
First mark sources with BYE and then schedule the BYE RTCP message.
 2013-07-26 12:17:58 +02:00
Wim Taymans
391943ba82 session: get SSRC from RTCP packet itself 
Get the SSRC from the RTCP packet instead.
 2013-07-26 12:17:57 +02:00
Wim Taymans
a3f75a17ef session: fix bandwidth calculation 
We iterate over all sources and the internal one is also in the
hashtable so avoid adding it twice.
 2013-07-26 12:17:57 +02:00
Wim Taymans
9eaef9d332 session: add some docs 2013-07-26 12:17:56 +02:00
Wim Taymans
2163355a47 session: Rearrange RTCP reporting a little 
Make a function to generate an RTCP packet for a source, pass the source as a
parameter.
Move timeout of collisions to session cleanup phase.
 2013-07-26 12:17:56 +02:00
Wim Taymans
a3bf374351 session: move check for is_early around 
Move the check for the early RTCP to where it is needed and used.
 2013-07-26 12:17:56 +02:00
Wim Taymans
b069db6a2e session: parse packet outside of the session lock 2013-07-26 12:17:56 +02:00
Wim Taymans
57c27ec319 session: do nicer checks for internal sources 2013-07-26 12:17:56 +02:00
Wim Taymans
93d07298ff session: let source keep track if it sent BYE 2013-07-26 12:17:56 +02:00
Wim Taymans
0c9c1434a8 source: reset more 2013-07-26 12:17:56 +02:00
Wim Taymans
1d02496d15 source: also use the source for bye_reason 
Store the BYE reason in our internal source object. Rename the methods on the
source object a little because now the BYE can be received in RTCP or
set when the session wants to send BYE.
 2013-07-26 12:17:56 +02:00
Wim Taymans
ddd071e54c session: configure sdes with structure only 
Remove code to configure the SDES with methods and types, only
allow configuration with GstStructure
 2013-07-26 12:17:55 +02:00
Wim Taymans
0060e1d45d session: refactor add and find source 
Make functions to find and add a source to the hashtable.
 2013-07-26 12:17:55 +02:00
Wim Taymans
adb0d68c07 session: remove source from sync_rtcp 
We don't need to know the sender source of the session in the
callback, the SR packet is for all participants in the session.
 2013-07-26 12:17:55 +02:00
Wim Taymans
bf7d8173b3 jitterbuffer: add some more debug 2013-07-26 12:17:55 +02:00
Vincent Penquerc'h
91d4abceaa aacparse: allow conversion from ADTS to raw AAC 
Some muxers (eg, qtmux) only support raw AAC, so this allows linking
an encoder that outputs ADTS only to those muxers.

The conversion is simple (omit the first 7 or 9 bytes of the frame),
but has to be done in pre_push instead of handle_frame as 1.0 does
not seem to allow skipping bytes there as 0.10 used to.

Other conversions are not supported (yet).
 2013-07-26 09:44:11 +01:00
Vincent Penquerc'h
55e9338846 aacparse: fix object_type parsing off-by-one in ADTS frame 
According to http://wiki.multimedia.cx/index.php?title=ADTS,
the value stored in ADTS headers is one less than the object
type of the AAC stream.

A look at ffmpeg shows it also adds 1 to the value read off
the ADTS header.

Note that this might break other things that happen to have
an inverse off by one to match the existing code.
 2013-07-26 09:44:10 +01:00
Thiago Santos
7eac4c7c03 avidemux: fix seqnum handling for seeks 
Use the same seqnum as the seek for flushes/segments that are
caused by the seek. Also do the same for segment events

Fixes #676242
 2013-07-25 15:24:31 -03:00
Thiago Santos
8bd12e12b3 matroskademux: fix seqnum handling for seeks 
Use the same seqnum as the seek for flushes/segments that are
caused by the seek. Also do the same for segment events

Fixes #676242
 2013-07-25 15:24:31 -03:00
Thiago Santos
e49b6e7c35 qtdemux: correctly handle seqnum for seeks and segments 
Use the same seqnum on messages and events for derived events.
Fixed for flushes / stream-start / segment after a seek, and segment
after a segment.

Fixes #676242
 2013-07-25 15:24:31 -03:00
Wim Taymans
c44a29bd53 bin: fix compilation 2013-07-24 14:17:45 +02:00
Wim Taymans
cc92ef1db2 vrawdepay: fix UYVP format 2013-07-24 12:42:31 +02:00
Wim Taymans
8191b6fcd2 vrawpay: fix UYVP format 2013-07-24 12:41:58 +02:00
Wim Taymans
37af93c361 vrawpay: fix caps 2013-07-24 12:41:44 +02:00
Wim Taymans
f87875e35b rtpjitterbuffer: fix locking 
Take the lock earlier so that we do things that follow with the right
locking.
 2013-07-24 10:49:03 +02:00
Wim Taymans
dece8413ef rtpsession: don't use invalid times in RTCP timeouts 
An invalid timeout can be calculated when we disabled RTCP by setting the
bandwidth to 0. Make sure all code can handle this case.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=674626
 2013-07-23 17:41:48 +02:00
Wim Taymans
25e0f0d6b6 rtpsession: lock session when changing bandwidth 
Take the session lock when changing the bandwidth properties so that we don't
end up with inconsistent behaviour.
 2013-07-23 17:41:48 +02:00
Wim Taymans
c337265ee4 session: reset some RTCP variables 
The early_send time was set to 0 and always triggering an early RTCP packet.
 2013-07-23 17:41:48 +02:00
Edward Hervey
3d48d25756 qtdemux: Add all the mpeg XDCAM variants 
This should cover all known XDCAM variants (which are all mpeg2 video)

Fixes #672227
 2013-07-23 15:03:31 +02:00
Carlos Rafael Giani
95429f1d4b rtpbin: added custom downstream sync event 
rtpbin can now send a custom in-band downstream event which informs
downstream that the bin has received an RTCP SR packet. This is useful
for applications which want to drop the initial unsynchronized received
RTP packets.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703560

Signed-off-by: Carlos Rafael Giani <dv@pseudoterminal.org>
 2013-07-23 06:25:20 +02:00
Tim-Philipp Müller
f18b1f7e80 deinterlace: fix on-the-fly changing of "mode" and "fields" properties 
We call setcaps() to reconfigure ourselves, but we need to pass
the current *sink* caps, not the source caps then. Also fix a
caps leak.

https://bugzilla.gnome.org/show_bug.cgi?id=641599
 2013-07-22 18:00:16 +01:00
Sebastian Dröge
0c2ff91a5c wavparse: Add support for group-id in the stream-start event 2013-07-22 15:30:13 +02:00
Sebastian Dröge
169b490664 rtspsrc: Add support for group-id in the stream-start event 2013-07-22 15:30:13 +02:00
Sebastian Dröge
5a9f4a3cbc rtpsession: Add support for group-id in the stream-start event 2013-07-22 15:30:13 +02:00