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Author SHA1 Message Date
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
Arnaud Vrac
e67655608c souphttpsrc: always ignore HEAD errors 
https://bugzilla.gnome.org/show_bug.cgi?id=704241
 2013-07-25 18:01:30 +02:00
Sebastian Dröge
ef4a21253b jpegenc: Clean up reset/start/stop handling 2013-07-25 14:26:37 +02:00
Sebastian Dröge
4944183061 jpegdec: Use base class error handling function instead of replicating it here 2013-07-25 14:26:37 +02:00
Sebastian Dröge
6f39f5d49f jpegdec: Clean up handling of reset/start/stop 2013-07-25 14:26:37 +02:00
Tim-Philipp Müller
a53dcf98f8 tests: fix test ID3 tags up not to rely on dodgy typefinding code 
Change 0xff 0xfb 'mp3' marker to 'fLaC' marker, so we can fix
the typefinder.

https://bugzilla.gnome.org/show_bug.cgi?id=681368
 2013-07-25 10:43:13 +01:00
Alessandro Decina
0df7af07ce osxaudiosink: intersect the probed caps with the filter passed to get_caps() 2013-07-25 08:24:45 +02:00
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
Sebastian Dröge
57dd1189d5 matroskademux: Add support for group-id in the stream-start event 2013-07-22 15:30:13 +02:00
Sebastian Dröge
1a0278ed64 qtdemux: Add support for group-id in the stream-start event 2013-07-22 15:30:13 +02:00
Sebastian Dröge
1122698491 flvdemux: Add support for group-id in the stream-start event 2013-07-22 15:30:12 +02:00
Sebastian Dröge
6cc16da531 avidemux: Add support for group-id in the stream-start event 2013-07-22 15:30:12 +02:00
Sebastian Dröge
c123519fed dvdemux: Add support for group-id in the stream-start event 2013-07-22 15:30:12 +02:00
Mathieu Duponchelle
d67a671bfb videomixer: use gst_util_uint64_scale*_round. 
There could be a case where:
      1) you do a new set_caps after buffers have been processed.
      2) ts_offset gets set to a different value, eg 0.033333333
      3) your pads get EOS, but the check dor that doesn't work
         because you use ts_offset + a truncated value < segment.stop
      4) so in the next collected, you end up comparing for example:
      0.9999999999 > 1., which is false and means you don't send EOS.

Also adds scale_round in two other places where it potentially could
have caused problems.
 2013-07-21 19:21:57 -04:00
Olivier Crête
96a8fb92e2 qtdemux: Add WRLE support 2013-07-19 14:58:30 -04:00
Tim-Philipp Müller
aa7d597120 qtdemux: make files from Vivotek camera play 
Skip tracks of 'vivo' subtype with empty stsd instead of
erroring out saying that the file is broken.

https://bugzilla.gnome.org/show_bug.cgi?id=699791
 2013-07-19 19:38:30 +01:00
Tim-Philipp Müller
ce52b319ff qtmux: when streaming don't try to seek when stopping 
It might cause errors in sinks that are not seekable and
have reported this (like e.g. fdsink)

https://bugzilla.gnome.org/show_bug.cgi?id=696228
 2013-07-19 17:31:38 +01:00
Wim Taymans
bdd3c31902 qtdemux: simplify some helpers 
Some helper functions are not needed anymore or can be simplified.
 2013-07-19 17:26:54 +02:00