Commit graph

13300 commits

Author SHA1 Message Date
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
Sebastian Rasmussen
35bb1b3328 docs: Add annotations for return values
Rephrase and clarify some return value descriptions

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725521
2014-03-02 23:41:18 +00:00
Sebastian Rasmussen
5b4f2ba20b docs: Fix argument and annotation typos
* colorbalance: Fix misspelled annotation
 * rtsp: Replace incorrectly documented function argument
 * sdp: Escape @ character to avoid gtk-doc warning
 * video-*: Add missing annotation colon
 * videodecoder/video-color: Fix function argument typos
 * videoutils: Remove unknown annotation field

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725521
2014-03-02 23:22:51 +00:00
Sebastian Rasmussen
53aab8eefa .gitignore: Ignore gcov intermediate files
https://bugzilla.gnome.org/show_bug.cgi?id=725479
2014-03-02 23:08:52 +00:00
Sebastian Dröge
e0b4f88c29 Automatic update of common submodule
From fe1672e to bcb1518
2014-02-28 09:34:31 +01:00
Matthieu Bouron
45dfceacdb playbin: improve autoplug_query_caps return
Makes autoplug_query_caps return
downstream_caps + intersect_first(filter_caps, element_caps)

https://bugzilla.gnome.org/show_bug.cgi?id=724828
2014-02-27 21:23:12 +01:00
Stefan Sauer
12bafecc6c Automatic update of common submodule
From 1a07da9 to fe1672e
2014-02-26 22:11:01 +01:00
Tim-Philipp Müller
14b82bbc9a rtsp: fix build with older GLib versions
The gio/gnetworking.h header is only available since glib 2.36

https://bugzilla.gnome.org/show_bug.cgi?id=725206
2014-02-26 11:44:18 +00:00
Ognyan Tonchev
5445682c6a rtspconnection: Add missing include
https://bugzilla.gnome.org/show_bug.cgi?id=725206
2014-02-26 11:25:13 +00:00
Matthieu Bouron
5c1167a2c7 playsinkconvertbin: improve gst_play_sink_convert_bin_getcaps return
If we have the peer caps and a caps filter, return peer_caps +
intersect_first (filter, converter_caps) instead of
intersect_first (filter, peer_caps + converter_caps) and preservers
downstream caps preference order.

https://bugzilla.gnome.org/show_bug.cgi?id=724893
2014-02-26 09:35:04 +01:00
Sebastian Rasmussen
ba9e8f0797 tests: Refactor RTP basepayloading test into pay/depay parts
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723328
2014-02-24 12:12:18 +01:00
Sebastian Rasmussen
d6dc1b6c46 rtpbasepayload: Let caps event also configure seqnum-offset
Previously the sequence number kept track of by GstRTPBasePayload would
only be set when going from READY to PAUSED state. This meant that a
downstream element that attempted to configure a basepayloader by
setting seqnum-offset e.g. in its sinkpad's caps template would have
trouble configuring the basepayloader. The reason was that the caps
event which arrives with the desired value for seqnum-offset did not
arrive at the basepayloader until caps negotiation took place,
significantly later than the transition from READY to PAUSED.

The result after this patch is that the default value for the
seqnum-offset property, or later set values for this property, will take
effect when going from READY to PAUSED like before. In addition the an
arriving caps event will also affect the basepayloaders configured
sequence number as the event arrives.
2014-02-24 12:10:26 +01:00
Sebastian Rasmussen
638d069c91 rtpbasepayload: Fix payload type property boundary value
The payload type field in an RTP packet header is 7 bits wide, hence the
boundary values ought to be 0x00 and 0x7f, not the previously stated
values 0x00 and 0x80.
2014-02-24 12:10:26 +01:00
Sebastian Rasmussen
3cc67ff494 rtpbasedepayload: Fix typos in comments 2014-02-24 12:10:26 +01:00
Tim-Philipp Müller
6442e76e9f docs: add GstVideoPool to docs 2014-02-23 14:42:12 +00:00
Sebastian Dröge
2df1e56bb7 decodebin: If we have a demuxer without dynamic srcpads, just assume no-more-pads
Otherwise we will wait until the multiqueue after the demuxer will
overrun, which is clearly not needed then.
2014-02-23 00:10:01 +01:00
Sebastian Dröge
f149c27a61 decodebin: Also make sure to not duplicate an element factory after a group
If we are using an adaptive stream demuxer, which outputs a non-container
stream, we are putting another multiqueue after the *parser* following
the adaptive stream demuxer. We do not want to add another instance of
the same parser right after this multiqueue.
2014-02-23 00:10:01 +01:00
Sebastian Dröge
41117606dd decodebin: During pre-rolling always use the auto-preroll limits on multiqueues
Even if we're buffering in the multiqueues.
2014-02-23 00:10:01 +01:00
Sebastian Dröge
2d2aa02b77 decodebin: Pass through the seekability information when setting multiqueue limits 2014-02-23 00:10:01 +01:00
Sebastian Dröge
db771185ed decodebin: During exposing of pads don't set the multiqueue limits multiple times to different values
Instead just set them once in the very end to the correct values.
2014-02-23 00:10:01 +01:00
Sebastian Dröge
c4caeb73ce decodebin: Only enable multiqueue buffering once we're pre-rolled
Otherwise we will emit buffering messages not just from the last
multiqueue but also from previous multiqueues... confusing the
application with different percentages during pre-rolling.
2014-02-23 00:10:01 +01:00
Sebastian Dröge
4f32010916 decodebin: Make sure that we always have a second multiqueue for adaptive streaming demuxers
For adaptive streaming demuxer we insert a multiqueue after
this demuxer. This multiqueue will get one fragment per buffer.
Now for the case where we have a container stream inside these
buffers, another demuxer will be plugged and after this second
demuxer there will be a second multiqueue. This second multiqueue
will get smaller buffers and will be the one emitting buffering
messages.
If we don't have a container stream inside the fragment buffers,
we'll insert a multiqueue below right after the next element after
the adaptive streaming demuxer. This is going to be a parser or
decoder, and will output smaller buffers.
2014-02-23 00:10:00 +01:00
Sebastian Dröge
ad51b38b7c uridecodebin: Always use buffering in multiqueue for adaptive streams 2014-02-23 00:10:00 +01:00
Sebastian Dröge
a2837a22a5 uridecodebin: Only add a queue2 for buffering for non-adaptive streaming streams 2014-02-23 00:10:00 +01:00
Thiago Santos
89c9e23bfe uridecodebin: pass on the buffering property for adaptive streams
Adaptive streams should download its data inside the demuxer, so
we want to use multiqueue's buffering messages to control the
pipeline flow and avoid losing sync if download rates are low;

https://bugzilla.gnome.org/show_bug.cgi?id=707636
2014-02-23 00:10:00 +01:00
Tim-Philipp Müller
3afd5fb3ab tests: add new unit tests to .gitignore 2014-02-21 19:07:59 +00:00
Ognyan Tonchev
6bf215fa09 rtspconnection: New unit test
See https://bugzilla.gnome.org/show_bug.cgi?id=724720
2014-02-21 16:21:45 +01:00
Ognyan Tonchev
ebe3530f51 rtspconnection: Remove read child source when POST is disconnected
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724720
2014-02-21 16:21:45 +01:00
Aleix Conchillo Flaqué
e8d58b432e defs: update for new rtspconnection symbols 2014-02-20 20:06:37 +01:00
Thiago Santos
b0985365af oggdemux: allow file to go until the end in push mode
When seeking back to original state after duration seeks, let
upstream know that we want the whole file, including the last
byte that wasn't requested on the duration seeks.

https://bugzilla.gnome.org/show_bug.cgi?id=724633
2014-02-20 00:24:02 -03:00
Thiago Santos
ef547c3eb5 oggdemux: remove unused instance variable event
It is never set to anything
2014-02-20 00:23:49 -03:00
Aleix Conchillo Flaqué
0a115bd31f rtspconnection: allow specifying a certificate database
Two new functions have been added,
gst_rtsp_connection_set_tls_database() and
gst_rtsp_connection_get_tls_database(). The certificate database will be
used when a certificate can't be verified with the default database.

https://bugzilla.gnome.org/show_bug.cgi?id=724393
2014-02-19 21:48:13 +01:00