Commit graph

13431 commits

Author SHA1 Message Date
Sebastian Dröge
2c2e286c38 decodebin: In adaptive streaming mode, only have a fixed buffer limit for the non-buffering multiqueue 2014-04-09 16:06:06 +02:00
Wim Taymans
4a81605d02 sdp: guard against address parse errors. 2014-04-08 15:59:47 +02:00
Mathieu Duponchelle
6954d2167c adder: rework the logic to check if eos has to be sent.
Checking the size available was incorrect, and the infos
for per-pad EOS are available.

Same logic as audiomixer.

fixes: https://bugzilla.gnome.org/show_bug.cgi?id=727025
2014-04-08 13:48:27 +02:00
Josep Torra
6ce7ade7c6 audioringbuffer: parse channels field from compressed audio caps
Also parse channels as an optional field in the caps for compressed
audio formats.
2014-04-08 12:54:04 +02:00
Jan Schmidt
968e28a818 subtitleoverlay: Consider all caps for overlays, not just the first.
Check all supported caps on the overlay video pad, not just the
first of (possibly) many.
2014-04-06 22:28:27 +10:00
Tim-Philipp Müller
b04675a1dc tools: update gst-play-1.0 man page 2014-04-05 13:25:46 +01:00
Thiago Santos
05e957106f videodecoder: do not deactivate the bufferpool, just unref
Videodecoder does late renegotiation, it will wait for the next
buffer before renegotiating its caps and bufferpool. It might happen
that downstream element switched from passthrough to non-passthrough
and sent a reconfigure upstream (that caused this renegotiation).
This downstream element will ask the video sink below for the bufferpool
with an allocation query and will get the same bufferpool that
videodecoder is holding, too.

When renegotiating, if videodecoder deactivates its bufferpool it
might be deactivating the bufferpool that some element downstream
is using and cause the pipeline to fail.

https://bugzilla.gnome.org/show_bug.cgi?id=727498
2014-04-04 13:50:03 -03:00
Vincent Penquerc'h
169166d0a2 audiobasesink: clip start samples to match clipped start time
Clock slaving can clip start time to zero, giving us a shorted
duration than we originally got. To keep in sync, we must then
discard the samples falling before that zero timestamp.

This possibly fixes random distortion caused by constant PA
underflows which are never resynced.
2014-04-04 17:04:06 +01:00
Wim Taymans
675d0400e1 mikey: Fix the KEMAC payload
The KEMAC payload actually needs to have subpayloads and the key should
go into the KEY_DATA subpayload. Add support for subpayloads and
implement the KEY_DATA payload.
Add some pointers to the conversion functions that allow us to add
encryption and decryption later.
2014-04-04 17:40:58 +02:00
Jan Schmidt
98482c3a0e playbin: Drop reference to any source element in NULL state
Drop the reference instead of waiting for either finalize(), or
for a new source when reused. Everyone else already forgot about
the old source.
2014-04-04 02:15:53 +11:00
Göran Jönsson
a483e90955 rtspconnection: Added gst_rtsp_watch_set_flushing to list.
Added gst_rtsp_watch_set_flushing to list in file
libgstrtsp.def
2014-04-03 13:30:33 +02:00
Sebastian Dröge
6189847ed0 videodecoder: Always drain the decoder after a discont group in reverse playback mode 2014-03-30 18:26:59 +02:00
Sebastian Dröge
5a4fbb1638 videodecoder: Flush the decoder once per discont group, not once per keyframe 2014-03-30 18:00:53 +02:00
Sebastian Dröge
f1f8731ff5 videodecoder: Handle reverse playback with multiple GOPs per discont group properly
baseparse will reverse each GOP for us already, so the segment events can
be after our keyframe. Make sure to get it and all other relevant sticky
events before starting to decode.
2014-03-30 17:59:55 +02:00
Sebastian Dröge
50c2218d4d videodecoder: Log event types of events that are pushed downstream 2014-03-29 10:33:01 +01:00
Sebastian Dröge
1c26e5734c videodecoder: In reverse playback mode we need to finish the subclass after passing all frames to it 2014-03-29 10:33:01 +01:00
Wim Taymans
8d439edd7a rtspconnection: add flush method
Add a method to set/unset the flushing state that makes _wait_backlog()
unlock.

See https://bugzilla.gnome.org/show_bug.cgi?id=725898
2014-03-28 09:34:33 +01:00
Nicolas Dufresne
f270b267c5 ximagesink: only extrapolate alpha mask for 32-bit depth
Instead of passing bogus alpha mask values when there's no alpha.

https://bugzilla.gnome.org/show_bug.cgi?id=727188
2014-03-27 16:47:30 -04:00
Wim Taymans
0348ee66f1 mikey: fix return values of g_return_* 2014-03-25 11:14:51 +01:00
Wim Taymans
183e441d88 rtsptransport: UDP is also default for SAVP and AVPF 2014-03-25 11:07:34 +01:00
Wim Taymans
51ca0bdf7b docs: add MIKEY docs 2014-03-24 17:12:52 +01:00
Wim Taymans
83888d6b13 mikey: add MIKEY parsing helpers
MIKEY is defined in RFC 3830 and is used to exchange SRTP encryption
parameters between a sender and a receiver in a secure way.
This library implements a subset of the features, enough to implement
RFC 4567, using MIKEY in SDP and RTSP.
2014-03-24 17:12:52 +01:00
Ognyan Tonchev
d7857325c5 rtspconnection: Fix minor memory leaks in error handling
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726642
2014-03-24 12:45:14 +01:00
Ognyan Tonchev
e0af857445 rtspconnection: Fix connection_poll()
* Only check for conditions we are interested in.
* Makes no sense to specify G_IO_ERR and G_IO_HUP in condition, they
  will always be reported if they are true.
* Do not create timed source if timeout is NULL.
* Correctly wait for sources to be dispatched, context_iteration() is
  not guaranteed to always block even if set to do so.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726641
2014-03-24 12:43:38 +01:00
Wim Taymans
bf4079277d rtpbasepayload: add pt and ssrc to stats 2014-03-20 09:19:46 +01:00
Thiago Santos
b7cf2fa571 tests: decodebin: port old decodebin2 test for parser and decoder linking
They were in the old decodebin2.c tests file and were never ported.
Now we can get rid of decodebin2.c
2014-03-16 14:36:51 -03:00
Arun Raghavan
f4cab18ec1 playback: Add video-/audio-filter properties
This provides an audio-filter and video-filter property to allow
applications to set filter elements/bins. The idea is that these will
e
applied if possible -- for non-raw sinks, the filters will be skipped.

If the application wishes to force the application of the filters, this
can be done by setting the new flag introduced on playsink -
GST_PLAY_FLAG_FORCE_FILTERS.

https://bugzilla.gnome.org/show_bug.cgi?id=679031
2014-03-16 18:38:29 +01:00
Sebastian Dröge
1bda077374 Revert "playback: Add video-/audio-filter properties"
This reverts commit fb8fdedb4f.
2014-03-16 18:38:25 +01:00
Arun Raghavan
fb8fdedb4f playback: Add video-/audio-filter properties
This provides an audio-filter and video-filter property to allow
applications to set filter elements/bins. The idea is that these will be
applied if possible -- for non-raw sinks, the filters will be skipped.

If the application wishes to force the application of the filters, this
can be done by setting the new flag introduced on playsink -
GST_PLAY_FLAG_FORCE_FILTERS.

https://bugzilla.gnome.org/show_bug.cgi?id=679031
2014-03-16 18:35:57 +01:00
Руслан Ижбулатов
d6bd37460a rtspconnection: Silence a compiler warning
Cast the argument into (const char *) on W32, as winsock2 expects it.

https://bugzilla.gnome.org/show_bug.cgi?id=726433
2014-03-16 11:22:04 +01:00
Arun Raghavan
bfb78cee42 playsink: Fix documentation for what the audio chain looks like
https://bugzilla.gnome.org/show_bug.cgi?id=679031
2014-03-16 11:02:43 +01:00
Tim-Philipp Müller
55b8e30a61 docs: update plugin docs and remove old properties and signals
Re-generate .args and .signals file from scratch so that
old signals that no longer exist (such as the 'new-decoded-pad'
signal on decodebin) no longer show up in the documentation.
2014-03-11 21:58:49 +00:00
Stefan Sauer
6cc7204f95 adder: set a group-id on the stream-start event
Set a default group-id to fix a warning printed by the sink.
2014-03-11 22:30:28 +01:00
Christian Fredrik Kalager Schaller
5648457e22 Add new header file 2014-03-11 17:40:17 +01:00
Thiago Santos
a2633b7cf1 oggmux: implement vp8 granulepos function
Add an extra function to the oggstream map to inform it about
the incoming buffers. This way oggmux can keep a count on the
vp8 invisible frames and calculate the granulepos correctly.

https://bugzilla.gnome.org/show_bug.cgi?id=722682
2014-03-11 12:28:49 -03:00
Thiago Santos
e00c306571 oggmux: create vp8 header data if not provided in caps
vp8 stream header shouldn't be assumed to be provided in caps always
as this would repeat the same code in all demuxers/encoders. Instead,
make oggmux generate them if they are not supplied.

https://bugzilla.gnome.org/show_bug.cgi?id=722682
2014-03-11 12:28:49 -03:00
Göran Jönsson
0b30fdbfbe rtspconnection: gst_rtsp_watch_wait_backlog
New method that wait until there is room in backlog queue.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
2014-03-10 17:28:40 +01:00
David Svensson Fors
6cd0d10d30 rtspconnection: GstRTSPWatch func for tunnel GET response
Add a callback in GstRTSPWatch where the response to HTTP GET for
tunneled connections can be modified.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725878
2014-03-10 10:43:03 +01:00
Wim Taymans
4898c30537 rtspdefs: add RFC 4567 headers and status code
This new Header and status code is used for SRTP
2014-03-10 10:33:28 +01:00
Sebastian Dröge
5e364c1d7b decodebin: Buffer up to 5 seconds in multiqueue buffering mode
2 seconds might be too small for some container formats, e.g.
MPEGTS with some video codec and AAC/ADTS audio with 700ms
long buffers. The video branch of multiqueue can run full while
the audio branch is completely empty, especially because there
are usually more queues downstream on the audio branch.
2014-03-07 17:09:24 +01:00
Sebastian Dröge
539eaf73e5 decodebin: Keep the number of buffers after an adaptive streaming demuxer lower
Usually these buffers are multiple seconds large, and having a maximum
of 5 buffers in the multiqueue there can use a lot of memory. Lower
this to 2 for adaptive streaming demuxers.
2014-03-06 22:45:30 +01:00
Sebastian Dröge
274b4eb870 decodebin: Simplify adaptive streaming demuxer code a bit 2014-03-06 22:45:30 +01:00
Adrien Schwartzentruber
a9d98c57a4 pango: demote debug WARNING to LOG for variable framerate video input
No need why we need to warn about that, it's perfectly allowed.

https://bugzilla.gnome.org/show_bug.cgi?id=725837
2014-03-06 17:51:11 +00:00
Matthieu Bouron
c904661dc3 tests: add textoverlay passthrough with composition feature unit tests
https://bugzilla.gnome.org/show_bug.cgi?id=721953
2014-03-05 20:39:01 +01:00
Matthieu Bouron
ed8e7d4275 pango: basetextoverlay: handle video/x-raw(ANY) if downstream supports the GstVideoOverlayCompositionMeta API
https://bugzilla.gnome.org/show_bug.cgi?id=721953
2014-03-05 20:38:53 +01:00
Matthieu Bouron
a8951c16da video-overlay-composition: add GST_CAPS_FEATURE_META_GST_VIDEO_OVERLAY_COMPOSITION 2014-03-05 20:38:45 +01:00
Andres Gomez
0e087b3865 docs: Removing GnomeVFS left bits
gnomevfs was removed time ago but there are still some left bits.

https://bugzilla.gnome.org/show_bug.cgi?id=725658
2014-03-05 20:25:39 +01:00
Tim-Philipp Müller
61fa4c7bb2 typefindfunctions: lower H.263 typefinder max probability
The typefinder returns LIKELY for as little as one possible
sync and no bad sync (not even taking into account how much
data was looked at for that). It's generally just not fit
for purpose, so should just not return anything like LIKELY
at all ever, even more so since it only recognises one out
of ten H263 files, and likes to mis-detect mp3s as H263.

https://bugzilla.gnome.org/show_bug.cgi?id=700770
https://bugzilla.gnome.org/show_bug.cgi?id=725644
2014-03-05 00:41:20 +00:00
Ognyan Tonchev
4220442441 rtspconnection: Call closed() when GET is closed in tunneled mode
This patch adds read source on the write socket in tunneled
mode and we get a callback when client disconnects the GET
channel.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725313
2014-03-03 10:34:56 +01:00
Sebastian Rasmussen
900c204eb9 videoformat: Remove duplicate/incorrect section
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725521
2014-03-02 23:41:51 +00:00