mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-26 03:31:05 +00:00
Code Issues 1 Projects Releases Wiki Activity
12972 commits 44 branches 1509 tags 396 MiB
Commit graph

6850 commits

Author SHA1 Message Date
Tim-Philipp Müller
47fccbe635 rtpdtmfsrc: fix compiler warning 
gstrtpdtmfsrc.c: In function 'gst_dtmf_src_prepare_message.isra.1':
gstrtpdtmfsrc.c:669:3: error: 's' may be used uninitialized in this function
 2013-01-26 22:58:29 +00:00
Olivier Crête
db5c3f4048 rtpdtmfdepay: Fix missing work in doc 2013-01-25 21:06:05 -05:00
Olivier Crête
92f9a9d9ff rtpdtmfsrc: Post the messages after the clock wait 
This way, the messages will be closer in time to when the packets are sent out
 2013-01-25 20:45:43 -05:00
Olivier Crête
0d316b4f43 rtpdtmfsrc: Only set the duration when starting to send 
The duration depends on the clock rate, which could change due to renegotiation
 2013-01-25 20:45:43 -05:00
Olivier Crête
90497aa3cd rtpdtmfsrc: remove "ssrc" from caps 
ssrc is uint and we don't have a uint range type
 2013-01-25 20:45:43 -05:00
Tim-Philipp Müller
d62019fff2 qtmux: set language to 'undefined' instead of English by default 2013-01-24 21:08:51 +00:00
Mark Nauwelaerts
0777a600e3 audioparsers: sbc: fix bogus compiler warning 
gst-plugins-good/gst/audioparsers/gstsbcparse.c: In function 'gst_sbc_parse_handle_frame':
gst-plugins-good/gst/audioparsers/gstsbcparse.c:210:32: error: 'ch_mode' may be used uninitialized i
 2013-01-22 19:26:09 +01:00
Thijs Vermeir
16128f0234 autoparsers: use appropriate printf format for gsize 2013-01-16 14:32:56 +01:00
Tim-Philipp Müller
9455a3aee1 rtpsbcpay: update some fields in the caps to their new name 
and to match the parser. "mode" got renamed to "channel-mode"
and "allocation" to "allocation-method".
 2013-01-16 10:19:36 +00:00
Tim-Philipp Müller
9f7a949773 audioparsers: add SBC audio parser 
From-scratch rewrite, the bluez one was useless and broken.

https://bugzilla.gnome.org/show_bug.cgi?id=690582
 2013-01-15 17:45:30 +00:00
Tim-Philipp Müller
39ef892938 rtp: import rtpsbcpay from bluez and port to 1.0 
Compiles, but not tested yet (sbc elements still need to be ported).

https://bugzilla.gnome.org/show_bug.cgi?id=690582
 2013-01-10 12:43:50 +00:00
Olivier Crête
c6dea5d09c dtmf/spandsp: Move dtmfdetect to use libspandsp 
Remove our copy of the tone_detect.c file and use the original
from libspandsp. Also move the element to the spandsp plugin.
 2013-01-09 20:05:16 -05:00
Marcel Holtmann
4196feb659 rtpsbcpay: Remove workaround for compiler warnings 2013-01-10 00:18:03 +00:00
Marcel Holtmann
fe79c60d74 rtpsbcpay: Add pragma based workaround for GStreamer warnings 2013-01-10 00:18:03 +00:00
Marcel Holtmann
08e95e7249 rtpsbcpay: Update copyright information 2013-01-10 00:15:36 +00:00
Marcel Holtmann
7fa03c0076 rtpsbcpay: Fix signed/unsigned comparison issue within GStreamer plugin 2013-01-10 00:15:35 +00:00
Marcel Holtmann
27a6b0abfe rtpsbcpay: Update copyright information 2013-01-10 00:15:35 +00:00
Marcel Holtmann
f890079aae rtpsbcpay: First attempt in fixing compiler warnings (still needs cleanup) 2013-01-10 00:15:35 +00:00
Johan Hedberg
7d4f846112 rtpsbcpay: More coding style fixes 2013-01-10 00:15:35 +00:00
Luiz Augusto von Dentz
151ad9b28d rtpsbcpay: Remove possible extra memcpy for gstreamer plugin. 2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz
69c8374b7c rtpsbcpay: Fix bug sending empty packages and remove a buffer copy. 2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz
7b3e4356ea rtpsbcpay: Fix runtime warnings of gstreamer plugin. 2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz
f74f061f3b rtpsbcpay: Update gstreamer plugin to use new sbc API. 2013-01-10 00:13:14 +00:00
Marcel Holtmann
b9be04f07b rtpsbcpay: Update copyright information 2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz
687400ecf4 rtpsbcpay: Fixes gstreamer caps and code cleanup. 2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz
a4f9624261 rtpsbcpay: Fix gtreamer payloader sending fragmented frames. 2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz
41e2f4f544 rtpsbcpay: Fix use of gstreamer plugin with rhythmbox and banshee and rtp timestamps. 2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz
96971cd323 rtpsbcpay: Make a2dpsink to act like a bin and split the payloader. 2013-01-10 00:13:14 +00:00
Wim Taymans
72402cc649 rtp: small improvements 2013-01-08 16:27:42 +01:00
Wim Taymans
af055d9574 jitterbuffer: refactor handle sync code 
Move the code that combines the last SR packet and the current jitterbuffer sync
values into a sync structure, into its own function. We want to reuse this bit
later.
 2013-01-07 15:50:33 +01:00
Wim Taymans
87f7d6b9bf rtp: include downstream latency in SR calculations 
When we make a mapping between an RTP timestamp and an NTP timestamp, include
the downstream latency applied to the sinks. This makes it possible to have
both sinks run with different latencies and still have correct sync on the
client. It also is more correct because the RTP timestamp in the SR report will
actually correspond more closely to the NTP time it was sent on the server.
For pipelines with high latency on the sender side, this actually allows a
GStreamer receiver to perform synchronisation instead of dropping the RTCP
packets.
 2013-01-07 15:45:10 +01:00
Wim Taymans
c631ed3300 rtpsession: don't cast event functions 
There is no need to cast the event functions and only causes problems later when
we change the signature later and things silently compiles wrong code.
 2013-01-07 14:25:14 +01:00
Wim Taymans
8dcde8b3ea rtp: more debug 2013-01-07 14:23:34 +01:00
Wim Taymans
6b7d05ac57 rtpsession: improve debug 2013-01-07 14:22:48 +01:00
Tim-Philipp Müller
cf1f6aff0d udpsrc: sanity check size of available packet data for reading to avoid memory waste 
On Windows and OS/X, _get_available_bytes() may not return the size
of the next pending packet, but the size of all pending packets in
the kernel-side buffer, which might be rather large depending on
configuration. Sanity-check the size returned by _get_available_bytes()
to make sure we never allocate more memory than the max. size for
a packet, if it's an IPv4 socket.

https://bugzilla.gnome.org/show_bug.cgi?id=610364
 2013-01-04 14:00:55 +00:00
Tim-Philipp Müller
95a37196b3 rtspsrc: add "proxy-id" and "proxy-pw" properties 
to match souphttpsrc. user/password passed via the URI
will still take precedence though.

https://bugzilla.gnome.org/show_bug.cgi?id=395427
 2012-12-31 00:22:27 +00:00
Wim Taymans
8cfec6a88d rtspsrc: fix cmd comparison 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=690476
 2012-12-20 17:12:30 +01:00
Wim Taymans
75616fac9a rtspsrc: add some more debug 2012-12-20 17:12:20 +01:00
Jonas Holmberg
e12457f138 rtpjpegpay: handle width and height > 2040 
If width or height is greater than 2040 set width and height to zero in
the rtp header and add x-dimensions to outcaps.

Solves #684955
 2012-12-20 15:40:49 +01:00
Wim Taymans
dcb0e0af93 avidemux: cleanup in flag define 2012-12-20 13:04:52 +01:00
Wim Taymans
0e522bc69a avidemux: improve debug 2012-12-20 13:04:52 +01:00
Thijs Vermeir
de41376231 rtp: use appropriate printf format for gsize 2012-12-18 16:02:09 +01:00
Thijs Vermeir
df88341ffb deinterlace: use appropriate printf format for gsize 2012-12-18 16:02:09 +01:00
Philippe Normand
2bd77e1c8a interleave: set src pad caps upon last sink pad CAPS event 
Gather caps on all sink pads before setting the src pad caps. This is
specially needed when the audio channel mapping is set on the sink
pads and the element needs to preserve it on its src pad.

https://bugzilla.gnome.org/show_bug.cgi?id=690267
 2012-12-18 12:58:43 +01:00
Tim-Philipp Müller
f4cb0c4315 matroskademux: skip empty tags 
instead of trying to add tags with empty strings, which
causes criticals at runtime.

https://bugzilla.gnome.org/show_bug.cgi?id=690358
 2012-12-17 22:55:12 +00:00
Sebastian Dröge
c49dede772 audioparsers: Make sure the caps are actually writable before changing them 2012-12-17 15:17:12 +01:00
Sebastian Dröge
26040ee38c audioparsers: Use the peer caps for restrictions instead of the srcpad allowed caps 
Otherwise we will intersect with the srcpad template caps and add all the caps fields
that the parser will ever set, no matter if downstream restricts this field or not.
This requires upstream to set this field on the caps to successfully negotiate.

https://bugzilla.gnome.org/show_bug.cgi?id=690184
 2012-12-17 15:01:02 +01:00
Alexey Fisher
7e47e3b92d matroskamux: set appropriate block header flag for VP8 invisible frames 
Useful for debugging mostly.

https://bugzilla.gnome.org/show_bug.cgi?id=654259
 2012-12-16 23:30:13 +00:00
Tim-Philipp Müller
8a3b116d1f docs: add rtpmux and rtpdtmfmux to plugin docs 
https://bugzilla.gnome.org/show_bug.cgi?id=629117
 2012-12-16 16:36:39 +00:00
Tim-Philipp Müller
3295b5d791 rtpmanager: move rtpmux and rtpdtmfmux elements from -bad 
https://bugzilla.gnome.org/show_bug.cgi?id=629117
 2012-12-16 16:36:39 +00:00
Tim-Philipp Müller
de204ba754 rtpmux: Fix FSF address 
https://bugzilla.gnome.org/show_bug.cgi?id=687520
 2012-12-16 16:36:39 +00:00
Tim-Philipp Müller
2778a1757f rtpmux: Use gst_element_class_set_static_metadata() 
where possible. Avoids some string copies. Also re-indent
some stuff. Also some indent fixes here and there.
 2012-12-16 16:36:39 +00:00
Olivier Crête
15dfdc58d4 rtpmux: Misc fix for 0.11 
Convert the incoming caps before proxying them
Clear the last_pad when going to ready

tests: Implement accept_caps, don't leak event
 2012-12-16 16:36:38 +00:00
Wim Taymans
83262be703 rtpmux: update for RTP buffer api changes 2012-12-16 16:36:38 +00:00
Sebastian Dröge
f17064a8ea rtpmux: Update for GST_PLUGIN_DEFINE() API changes 2012-12-16 16:36:34 +00:00
Wim Taymans
c86156ad8f rtpmux: fix compilation 2012-12-16 16:35:36 +00:00
Wim Taymans
6826bbb6da rtpmux: fix for caps api changes 2012-12-16 16:35:33 +00:00
Matej Knopp
bb345a584d rtpmux: Fix compiler warnings 2012-12-16 16:35:29 +00:00
Olivier Crête
af4e999c59 rtpmux: Unref non-forwarded events 
Also, don't unref forwarded ones
 2012-12-16 16:35:29 +00:00
Olivier Crête
a8789d1df1 rtpmux: resync iterator on resync 2012-12-16 16:35:29 +00:00
Olivier Crête
0c54079af5 rtpmux: Re-push sticky events on input pad change 2012-12-16 16:35:29 +00:00
Olivier Crête
21831b430f rtpmux: Don't leak gvalue from iterator 2012-12-16 16:35:29 +00:00
Wim Taymans
ccc4b960fc rtpmux: more porting 2012-12-16 16:35:26 +00:00
Olivier Crête
f20a6b1d16 rtpmux: port to 0.11 2012-12-16 16:35:26 +00:00
Wim Taymans
35b6668fb6 rtpmux: make request pads take _%u 2012-12-16 16:35:22 +00:00
Olivier Crête
aa3607ef5c rtpdtmfmux: Add last-stop to dtmf-event upstream events 
Add the running time of the last outputted buffer to the
upstream "dtmf-event" events so that the dtmf source does not
leave a gap.
 2012-12-16 16:35:22 +00:00
Edward Hervey
d137482fe5 rtpmux: Remove dead assignments 2012-12-16 16:35:22 +00:00
Stefan Kost
55aae6bfab rtpmux: add missing G_PARAM_STATIC_STRINGS flags 
Canonicalize property names as needed.
 2012-12-16 16:35:15 +00:00
Olivier Crête
9674d5cc23 rtpmux: Improve documentation 
Add an example pipeline, and try to explain a bit more what it does.
 2012-12-16 16:35:15 +00:00
Stefan Kost
ca27a279ba rtpdtmfmux: remove unused variable 2012-12-16 16:35:15 +00:00
Stefan Kost
c85dceeacb rtpdtmfmux: remove unused signal boilerplate 2012-12-16 16:35:15 +00:00
Stefan Kost
2353f8d852 rtpmux: no need to ref pad in _chain() 2012-12-16 16:35:15 +00:00
Youness Alaoui
e42d2eebcb rtpmux: Unlock the right mutex 
The mutex locked is for the 'mux' object, but we unlock the
pad, which means that if the rtpmux gets a flush, then the
object lock will stay locked forever, causing it to freeze
the next time it tries to take it.

Fixes bug #627991
 2012-12-16 16:35:15 +00:00
Olivier Crête
78d1ebac9e rtpmux: Add support for GstBufferList 
Factor out most of the buffer handling and implement a chain_list
function. Also, the DTMF muxer has been modified to just have a
function to accept or reject a buffer instead of having to subclass
both chain and chain_list.
 2012-12-16 16:35:15 +00:00
Olivier Crête
c00f14419b rtpmux: Don't leak invalid buffers 2012-12-16 16:35:15 +00:00
Tim-Philipp Müller
a45429d81d rtpmux: fix missing debug log message argument 2012-12-16 16:35:15 +00:00
Olivier Crête
4a8d0243b5 rtpdtmfmux: Add some debug messages 2012-12-16 16:35:14 +00:00
Olivier Crête
423ce98666 rtpdtmfmux: Remove stream-lock event handling 2012-12-16 16:35:14 +00:00
Olivier Crête
a4500c0e74 rtpdtmfmux: Update doc for simplification 2012-12-16 16:35:14 +00:00
Olivier Crête
70097866de rtpdtmfmux: Drop buffers on non-priority sinks when something is incoming on the priority sink 2012-12-16 16:35:14 +00:00
Olivier Crête
f6548fe9b6 rtpdtmfmux: Add priority sink pads 2012-12-16 16:35:14 +00:00
Olivier Crête
2bcea1537b rtpdtmfmux: Cleanup event function 2012-12-16 16:35:14 +00:00
Olivier Crête
8e58646f5c rtpmux: Aggregate incoming segments 2012-12-16 16:35:14 +00:00
Olivier Crête
7be57cac3a rtpdtmfmux: Update documentation 2012-12-16 16:35:14 +00:00
Olivier Crête
e590fc1f32 rtpmux: Simplify request pad creation 2012-12-16 16:35:14 +00:00
Benjamin Otte
2867e00225 rtpmux: gst_element_class_set_details => gst_element_class_set_details_simple 2012-12-16 16:35:10 +00:00
unknown
fb7266884d rtpmux: update the current_ssrc from the caps 
Fixes #604101
 2012-12-16 16:33:47 +00:00
Håvard Graff
eab65e84ca rtpmux: release pads when disposing 
Because of an allocated priv (GstRTPMuxPadPrivate), the element will
leak memory if not gst_rtp_mux_release_pad() is called. This would
previously only happen if release_request_pad() was called explicitly,
somthing that should not be neccesary.

Fixes #604099
 2012-12-16 16:33:46 +00:00
Wim Taymans
0d54122804 dtmfmux: method name cleanups 2012-12-16 16:33:46 +00:00
Olivier Crête
3841cc74cf rtpmux: Don't ignore requested pad name 2012-12-16 16:33:46 +00:00
Olivier Crête
d93295ff9d rtpmux: Remove empty finalize 2012-12-16 16:33:46 +00:00
Olivier Crête
5e90a4e86b rtpmux: Free the pad private data on pad release 
Free the pad private data on pad release instead of using a weak ref,
which is not thread safe. Also, lock the content of the pad private using the element's
object lock.
 2012-12-16 16:33:46 +00:00
Olivier Crête
4be63c9add rtpmux: Reject wrong caps 2012-12-16 16:33:46 +00:00
Olivier Crête
0111bafb3a rtpmux: Fix leak Fixed a leak discovered by Laurent Glayal <spegle@yahoo.fr> 2012-12-16 16:33:46 +00:00
Olivier Crête
fcc1522d2e rtpmux: Fix leak 
Fixed a leak discovered by Laurent Glayal <spegle@yahoo.fr>
 2012-12-16 16:33:46 +00:00
Olivier Crête
ff6686f1c7 rtpmux: Fix warning 2012-12-16 16:33:46 +00:00
Olivier Crête
00791f930b rtpmux: Set different caps depending on the input 2012-12-16 16:33:46 +00:00
Olivier Crête
ed0b407038 rtpmux: Only free pad private when pad is disposed 2012-12-16 16:33:45 +00:00
Olivier Crête
92bb5199ac rtpmux: Remove useless caps mangling 2012-12-16 16:33:45 +00:00
Olivier Crête
3ccf3217fe rtpmux: Rename variable for more clarity 2012-12-16 16:33:45 +00:00
Olivier Crête
4b958f6d8d rtpmux: Use GST_BOILERPLATE 2012-12-16 16:33:45 +00:00
Olivier Crête
abe57be248 rtpmux: Do the includes locally 2012-12-16 16:33:45 +00:00
Olivier Crête
05844c89e9 rtpmux: Add GST_DEBUG_FUNCPTRs 2012-12-16 16:33:45 +00:00
Olivier Crête
fd102b95ab rtpdtmfmux: Release locked pad on release_pad 
Release the special pad if the pad is removed from the muxer.
 2012-12-16 16:33:45 +00:00
Laurent Glayal
00f8bab712 rtpdtmfmux: Release special on pad dispose 
Fixes #577690
 2012-12-16 16:33:45 +00:00
Stefan Kost
a4a22454dc docs: various doc fixes 
No short-desc as we have them in the element details.
Also keep things (Makefile.am and sections.txt) sorted.
Reword ambigous returns. No text after since please.
 2012-12-16 16:33:41 +00:00
Olivier Crête
7d4395a910 rtpmux: Move rtpmux from gst-plugins-farsight to -bad 2012-12-16 16:33:27 +00:00
Olivier Crête
68215752f4 rtpmux: Re-indent to Gst style 2012-12-16 16:33:24 +00:00
Olivier Crête
c7d0809434 rtpmux: Document rtp muxer a bit 2012-12-16 16:33:20 +00:00
Laurent Glayal
47c7a93df2 rtpmux: Add signals before stream lock and after unlocking 2012-12-16 16:33:17 +00:00
Olivier Crête
f1656ed8b0 rtpmux: Let ssrc through getcaps 2012-12-16 16:33:14 +00:00
Olivier Crête
1529dffaf9 rtpmux: Rename have_base to have_ts_base 2012-12-16 16:33:11 +00:00
Olivier Crête
57563517bd rtpmux: Protect the seqnum with object lock in rtpmux 2012-12-16 16:33:08 +00:00
Olivier Crête
d3237eaf95 rtpmux: Remove unused sink_ts_base 2012-12-16 16:33:04 +00:00
Olivier Crête
cc23958183 rtpmux: Have getcaps to force the same clockrate on all pads 2012-12-16 16:33:01 +00:00
Olivier Crête
dc36590d0c rtpmux: Validate RTP data in RTP Mux 2012-12-16 16:32:57 +00:00
Olivier Crête
360c8d4f1d rtpmux: Remove unused clock-rate property 2012-12-16 16:32:54 +00:00
Olivier Crête
b86232d0dc rtpmux: Clarify locking in rtpdtmfmux 2012-12-16 16:32:50 +00:00
Laurent Glayal
4b607cdda5 rtpmux: Missing format parameter 2012-12-16 16:32:47 +00:00
Håvard Graff
b313c80367 rtpmux: Update seqnum base in rtp muxer 
With help from Wim
 2012-12-16 16:32:43 +00:00
Håvard Graff
c479f90274 rtpmux: Fix some more leaks 2012-12-16 16:32:40 +00:00
Håvard Graff
1b5e769e0b rtpmux: Fix leak 2012-12-16 16:32:37 +00:00
Olivier Crête
5cbb0de823 rtpmux: Don't unref caps we don't know (thanks Wim) 2012-12-16 16:32:32 +00:00
Olivier Crête
cebf506949 rtpmux: Put per-buffer debug at level LOG 2012-12-16 16:32:29 +00:00
Olivier Crête
3c12a423b7 rtpmux: Make debug print accurate 2012-12-16 16:32:25 +00:00
Olivier Crête
c49f4c87c6 rtpmux: Set our caps on the buffers 2012-12-16 16:32:22 +00:00
Olivier Crête
ec63da9366 rtpmux: Take the clock-base stored from the last setcaps 2012-12-16 16:32:18 +00:00
Olivier Crête
674c074114 rtpmux: Store the clock-base on setcaps 2012-12-16 16:32:15 +00:00
Olivier Crête
90264b9686 rtpmux: Add padprivate to the request pads 2012-12-16 16:32:11 +00:00
Olivier Crête
15d661ba3e rtpmux: Make indentation more correct 2012-12-16 16:31:56 +00:00
Olivier Crête
3a7d09a749 rtpmux: Fix typo 2012-12-16 16:31:53 +00:00
Olivier Crête
91aef3ec5e rtpmux: Set seqnum-base and clock-base in caps from rtpmuxer 2012-12-16 16:31:50 +00:00
Zeeshan Ali
6ea5ca354d rtpmux: more debug 
20070815135038-f3f1e-9c7a5490a525c6e8753cb1b8c03354df99132b5c.gz
 2012-12-16 16:31:46 +00:00
Youness Alaoui
f0e209b638 rtpmux: missing comment 
20070820185032-4f0f6-0ab67b6ac40dd4e35a8fe53f3cb6daff65ce43b9.gz
 2012-12-16 16:30:33 +00:00
Olivier Crete
3ed5590da6 rtpmux: Make buffer writable before writing into it 
20070712195336-3e2dc-91a5fb797cfa4919d4e2f9a728c6d6fbd3b83d93.gz
 2012-12-16 16:30:31 +00:00
Olivier Crete
dd13f7c8ef rtpmux: Set pads active when adding them to a potentially running element 
20070706202459-3e2dc-a3731f885725594def0a7be997fc7b3a739ee967.gz
 2012-12-16 16:30:27 +00:00
Olivier Crete
1c5075f927 rtpmux: Fix multiple ref leaks (patches by SP GLE) 
20070607120121-3e2dc-061e9ef7a47b1b84fa8f8092f4b8bcc0e6db8c8c.gz
 2012-12-16 16:30:23 +00:00
Zeeshan Ali
42f455e902 rtpmux: send event to all src pads 
20070528152505-f3f1e-039216c73dc93f64c49962c77a0253cb9cfec4d3.gz
 2012-12-16 16:30:18 +00:00
Zeeshan Ali
dba101bb0f rtpmux: print a warning if receive an error iterating sinkpads 
20070528123749-f3f1e-4c1eb3f511b5610143610a65a94d117f2c3d2580.gz
 2012-12-16 16:30:15 +00:00
Zeeshan Ali
baa48dc6bc rtpmux: deal with all the gst_iterator_next() return values 
20070528122808-f3f1e-d301644c3be7633ec6dc5e28596e9346d2da6a50.gz
 2012-12-16 16:30:12 +00:00
Zeeshan Ali
de40874670 rtpmux: Return correct value from the event handler 
20070525123116-f3f1e-131b37b5f4521618fe2f1320409a47e65b35ad2d.gz
 2012-12-16 16:30:08 +00:00
Zeeshan Ali
ed76f67e96 rtpmux: Ville's original patch to fix the traversal of dtmf event 
20070525102709-f3f1e-6c41d1ef934068a4f4e810e7e981b420075b0c98.gz
 2012-12-16 16:30:05 +00:00
zeeshan.ali@nokia.com
94ebe07862 rtpmux: Set the correct ts-offset on the get_prop value 
20070329135250-65035-a43e222d91d57c0a61cb3287586aaa29abf78674.gz
 2012-12-16 16:30:01 +00:00
zeeshan.ali@nokia.com
1ee542c378 rtpmux: Refactorize state_change 
20070329135223-65035-23a0107b2e397710f035c6e88cc0e49b65bb4d5d.gz
 2012-12-16 16:29:58 +00:00
zeeshan.ali@nokia.com
2498ba671a rtpmux: set SSRC on the packets 
20070329133622-65035-1be6e0aa85a71389f7d257b9cd3e13a73d6b745b.gz
 2012-12-16 16:29:55 +00:00
zeeshan.ali@nokia.com
ee69c2690d rtpmux: Code clean-up and more debug output 
20070329131936-65035-9d499e209e0d7a409c3aa0d1040778babf076179.gz
 2012-12-16 16:29:52 +00:00
zeeshan.ali@nokia.com
1c799ce964 rtpmux: Use own clock-base 
20070328112219-65035-1ba5fefbc65059e9b0c860528a31062ceb6a7331.gz
 2012-12-16 16:29:48 +00:00
zeeshan.ali@nokia.com
b04630d7a2 rtpmux: Only accept RTP streams that have the same clock-rate 
20070323163139-65035-fc0b17b0b8a7a041f48994c4f26e96568168bf95.gz
 2012-12-16 16:29:45 +00:00
zeeshan.ali@nokia.com
6fe1e02efd rtpmux: Some more code-cleanups 
20070322161552-65035-bda96165e146b4f1d5fea1cc9576a7ab3abebc9e.gz
 2012-12-16 16:29:42 +00:00
zeeshan.ali@nokia.com
1603223ee5 rtpmux: return newpad instead of NULL and warn if failed to create a pad 
20070322154251-65035-cdb6651e61c2eb0205cc8c24693b43f98a2da718.gz
 2012-12-16 16:29:38 +00:00
zeeshan.ali@nokia.com
23d3ed5c5f rtpmux: Refactorize the RTPMux code 
20070322124132-65035-0a3278147546e33f687097a43b775b3f6aa99f93.gz
 2012-12-16 16:29:35 +00:00
zeeshan.ali@nokia.com
21e6e951f6 rtpmux: Some more doc fixing 
20070322121453-65035-12d602272217b51bd97df4e5790024c399622dd3.gz
 2012-12-16 16:29:32 +00:00
zeeshan.ali@nokia.com
0de7fb6f37 rtpmux: More Refactoring 
20070322113228-65035-bae34a79599e7de5293ed77b022361ccff822bb9.gz
 2012-12-16 16:29:29 +00:00
zeeshan.ali@nokia.com
0f755657ce rtpmux: More documentation 
20070322113154-65035-624850541a5b5fc3df231204be5a83d07239db28.gz
 2012-12-16 16:29:26 +00:00
zeeshan.ali@nokia.com
5483c78ac0 rtpmux: Refactor the event handler function 
20070321163311-65035-987e7f25d1ab5335b79f44b277abf15e4e37d317.gz
 2012-12-16 16:29:23 +00:00
zeeshan.ali@nokia.com
db1523ae60 rtpmux: Add RTPDTMFMux element 
20070321145244-65035-9a01390b0dee3398e53199a1fa1d9352004f338e.gz
 2012-12-16 16:29:19 +00:00
zeeshan.ali@nokia.com
97ff54dce7 rtpmux: Remove DTMF-specific code from RTP muxer and make it extendable 
20070321123149-65035-b8a8f55ff78eed8cbb0042e827885edfc5438242.gz
 2012-12-16 16:29:16 +00:00
zeeshan.ali@nokia.com
1a227ac7e5 rtpmux: Put more helpful description 
20070320120524-65035-db27a7cf6307b511aeb3d996d26e790e367a7bad.gz
 2012-12-16 16:29:13 +00:00
zeeshan.ali@nokia.com
d876c0d8cc rtpmux: remove the (commented-out) code for blocking the pads 
20070316151641-65035-0123af387951f88594797c722e882cfe70240aff.gz
 2012-12-16 16:29:10 +00:00
zeeshan.ali@nokia.com
209228c44d rtpmux: Drop buffers instead of blocking the sinkpads 
20070316131444-65035-9c1345ad96108881f455d4b55a7f623cd302d0ed.gz
 2012-12-16 16:29:05 +00:00
zeeshan.ali@nokia.com
795822ffa5 rtpmux: Implement stream locking, needed for DTMF 
20070314171618-65035-e4d24b1606ce0a3e2e739f01833f61e4d7555eac.gz
 2012-12-16 16:29:02 +00:00
zeeshan.ali@nokia.com
fd209faa56 rtpmux: use GST_*_OBJECT instead of g_* 
20070314102058-65035-e2442888f2e3e5a3a7659ad7954a4fba34749ce2.gz
 2012-12-16 16:28:58 +00:00
zeeshan.ali@nokia.com
b0208cb0a6 rtpmux: No need to manage pads, parent does that for us 
20070314101854-65035-ef5f4abde227102a1128835ab325905eae4c3726.gz
 2012-12-16 16:28:55 +00:00
zeenix@gmail.com
74e9071dad rtpmux: Fix copyright header 
20070314090358-d014a-3a6d3eeeaaf5cb8ca3bca6a33e99a551f598bd48.gz
 2012-12-16 16:28:51 +00:00
zeeshan.ali@nokia.com
3c4cdf1541 rtpmux: The first implementation of RTP muxer 
20070307085307-65035-833402413f99cb3f8be4883e92bad4c8722510c9.gz
 2012-12-16 16:28:41 +00:00
Tim-Philipp Müller
b19122bac8 scaletempo: no need for a private struct 2012-12-15 21:27:01 +00:00
Tim-Philipp Müller
61913ab7b4 audiofx: move scaletempo element from -bad 
https://bugzilla.gnome.org/show_bug.cgi?id=687262
 2012-12-14 13:16:17 +00:00
Sebastian Dröge
314765c294 scaletempo: Fix event leak 2012-12-14 13:16:17 +00:00
Sebastian Dröge
490e408991 scaletempo: Fix timestamp tracking 2012-12-14 13:16:17 +00:00
Sebastian Dröge
502eb8d1b7 scaletempo: Implement LATENCY query 2012-12-14 13:16:17 +00:00
Sebastian Dröge
c7589817cb scaletempo: Store instance private data in the instance struct 
Getting it over and over again via G_TYPE_INSTANCE_GET_PRIVATE()
is really slow.
 2012-12-14 13:16:17 +00:00
Tim-Philipp Müller
e552bd484f scaletempo: use gst_element_class_set_static_metadata() 
where possible. Avoids some string copies. Also re-indent
some stuff. Also some indent fixes here and there.
 2012-12-14 13:16:17 +00:00
Mark Nauwelaerts
d2dd91ac47 scaletempo: replace gst_element_class_set_details_simple with gst_element_class_set_metadata 2012-12-14 13:16:17 +00:00
Wim Taymans
cb1743d578 scaletempo: ffmpegcolorspace is no more 2012-12-14 13:16:17 +00:00
Sebastian Dröge
93e1091d7f scaletempo: Update for GST_PLUGIN_DEFINE() API changes 2012-12-14 13:16:17 +00:00
Mark Nauwelaerts
3286cdd542 scaletempo: port to 0.11 2012-12-14 13:16:16 +00:00
Stefan Kost
62d780cd51 scaletempo: improve the docs 
Fix the syntax, add more explanation and xref the properties.
 2012-12-14 13:16:16 +00:00
Chris E Jones
caf2b6cb5c scaletempo: Correctly handle newsegment events with stop==-1 
Fixes bug #645420.
 2012-12-14 13:16:16 +00:00
Stefan Kost
6d54058982 scaletempo: add missing G_PARAM_STATIC_STRINGS flags 
Canonicalize property names as needed.
 2012-12-14 13:16:16 +00:00
Benjamin Otte
38bc2dfb4a scaletempo: gst_element_class_set_details => gst_element_class_set_details_simple 2012-12-14 13:16:16 +00:00
Thiago Santos
2d72ec153a scaletempo: properly update new segments 
Scaletempo was missing an update of 'stop' in
new segment parameters when pushing it downstream,
which caused files to end earlier when rate < 1.

Fixes #599903

Based on patch by: Bastian Hecht <hechtb@gmail.com>
 2012-12-14 13:16:16 +00:00
Maximilian Högner
2fe7a97f1c scaletempo: Explicitely cast to signed integers to fix a segfault 
Fixes bug #585660.
 2012-12-14 13:16:16 +00:00
Michael Smith
1b1f6f56d6 scaletempo: Do not use void pointer arithmetic. 2012-12-14 13:16:16 +00:00
Stefan Kost
9284c85b33 scaletempo: Return the result of parent_class->event() 
Original commit message from CVS:
* gst/audiofx/gstscaletempo.c:
Return the result of parent_class->event().
 2012-12-14 13:16:16 +00:00
Rov Juvano
43e79f7769 Add scaletempo plugin, which allows to scale the speed of audio without changing the pitch by handling seeks with a r... 
Original commit message from CVS:
Patch by: Rov Juvano <rovjuvano at users dot sourceforge dot net>
* configure.ac:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/inspect/plugin-scaletempo.xml:
* examples/scaletempo/Makefile.am:
* examples/scaletempo/demo-gui.c: (pop_status_bar),
(status_bar_printf), (demo_gui_seek_bar_format), (update_position),
(demo_gui_seek_bar_change), (demo_gui_do_change_rate),
(demo_gui_do_set_rate), (demo_gui_do_rate_entered),
(demo_gui_do_toggle_advanced), (demo_gui_do_toggle_disabled),
(demo_gui_do_seek), (demo_gui_do_play), (demo_gui_do_pause),
(demo_gui_do_play_pause), (demo_gui_do_open_file),
(demo_gui_do_playlist_prev), (demo_gui_do_playlist_next),
(demo_gui_do_about_dialog), (demo_gui_do_quit),
(demo_gui_request_set_stride), (demo_gui_request_set_overlap),
(demo_gui_request_set_search), (demo_gui_rate_changed),
(demo_gui_playing_started), (demo_gui_playing_paused),
(demo_gui_playing_ended), (demo_gui_player_errored),
(demo_gui_stride_changed), (demo_gui_overlap_changed),
(demo_gui_search_changed), (demo_gui_set_player_func),
(demo_gui_set_playlist_func), (build_gvalue_array),
(create_action), (demo_gui_show_func), (demo_gui_set_player),
(demo_gui_set_playlist), (demo_gui_show), (demo_gui_get_property),
(demo_gui_set_property), (demo_gui_init), (demo_gui_class_init),
(demo_gui_get_type):
* examples/scaletempo/demo-gui.h:
* examples/scaletempo/demo-main.c: (handle_error_message),
(handle_quit), (main):
* examples/scaletempo/demo-player.c: (no_pipeline),
(demo_player_event_listener), (demo_player_state_changed_cb),
(demo_player_eos_cb), (demo_player_build_pipeline), (_set_rate),
(demo_player_scale_rate_func), (demo_player_set_rate_func),
(_set_state_and_wait), (demo_player_load_uri_func),
(demo_player_play_func), (demo_player_pause_func), (_seek_to),
(demo_player_seek_by_func), (demo_player_seek_to_func),
(demo_player_get_position_func), (demo_player_get_duration_func),
(demo_player_scale_rate), (demo_player_set_rate),
(demo_player_load_uri), (demo_player_play), (demo_player_pause),
(demo_player_seek_by), (demo_player_seek_to),
(demo_player_get_position), (demo_player_get_duration),
(demo_player_get_property), (demo_player_set_property),
(demo_player_init), (demo_player_class_init),
(demo_player_get_type):
* examples/scaletempo/demo-player.h:
* gst/audiofx/Makefile.am:
* gst/audiofx/gstscaletempo.c: (best_overlap_offset_float),
(best_overlap_offset_s16), (output_overlap_float),
(output_overlap_s16), (fill_queue), (reinit_buffers),
(gst_scaletempo_transform), (gst_scaletempo_transform_size),
(gst_scaletempo_sink_event), (gst_scaletempo_set_caps),
(gst_scaletempo_get_property), (gst_scaletempo_set_property),
(gst_scaletempo_base_init), (gst_scaletempo_class_init),
(gst_scaletempo_init):
* gst/audiofx/gstscaletempo.h:
* gst/audiofx/gstscaletempoplugin.c: (plugin_init):
Add scaletempo plugin, which allows to scale the speed of audio without
changing the pitch by handling seeks with a rate!=1.0.
Integrate it into the docs and add the example application for it.
Fixes bug #537700.
 2012-12-14 13:16:15 +00:00
Havard Graff
9c94f1187c jitterbuffer: bundle together late lost-events 
The scenario where you have a gap in a steady flow of packets of
say 10 seconds (500 packets of with duration of 20ms), the jitterbuffer
will idle up until it receives the first buffer after the gap, but will
then go on to produce 499 lost-events, to "cover up" the gap.

Now this is obviously wrong, since the last possible time for the earliest
lost-events to be played out has obviously expired, but the fact that
the jitterbuffer has a "length", represented with its own latency combined
with the total latency downstream, allows for covering up at least some
of this gap.

So in the case of the "length" being 200ms, while having received packet
500, the jitterbuffer should still create a timeout for packet 491, which
will have its time expire at 10,02 seconds, specially since it might
actually arrive in time! But obviously, waiting for packet 100, that had
its time expire at 2 seconds, (remembering that the current time is 10)
is useless...

The patch will create one "big" lost-event for the first 490 packets,
and then go on to create single ones if they can reach their
playout deadline.

See https://bugzilla.gnome.org/show_bug.cgi?id=667838
 2012-12-13 12:00:43 +01:00
Wim Taymans
a858bf46db rtspsrc: fix TCP reconnect 
Ignore other commands when reconnecting, otherwise the loop function would pause
and the reconnection would not happen. Continue looping after doing a reconnect
so that we have a chance to actually read the new data.
 2012-12-13 09:30:59 +01:00
Philippe Normand
a8fa9f2b47 deinterleave: properly set srcpad channel position 
The src pad caps always describe a single audio channel so only the
first position matters if deinterleave is configured to keep channel
positions in its src pads.
 2012-12-12 11:20:56 +00:00
Wim Taymans
b1dc816772 rtspsrc: timeout on udpsrc is in nanoseconds 2012-12-12 11:09:42 +01:00
Wim Taymans
32bd981303 udpsrc: improve timeouts 
Make it possible to set the timeout after we went to the READY state by using
the timeout when checking the condition. This also makes it possible to set the
timeout with a higher granularity than seconds.
 2012-12-12 11:08:13 +01:00
Wim Taymans
abd7e33db6 deinterlace: add support for strides 
Implement stride support correctly by taking it from the GstVideoFrame.
Propose a bufferpool upstream when not operating in passthrough.
 2012-12-11 13:00:46 +01:00
Aleix Conchillo Flaque
3503aef946 rtspsrc: do not change state to PLAYING if currently chaning state 
* gst/rtsp/gstrtspsrc.c (gst_rtspsrc_play): state change might be
  happening in the application thread, so we don't change the state to
  PLAYING in the gstrtspsrc thread unless it is safe.

  A specific case is when chaning the state to NULL from the application
  thread. This will synchronously try to stop the task (with the element
  state lock acquired), but we will try a gst_element_set_state from
  gstrtspsrc thread which will block on the element state lock causing a
  deadlock.

  https://bugzilla.gnome.org/show_bug.cgi?id=684312
 2012-12-10 15:13:22 +01:00
Tim-Philipp Müller
672ab8fb5b webmux: fix linking with shout2send element 
Shout2send only accepts webm format, not matroska, but due
to a bug in matroskamux, webmmux's source pad is also created
with the matroska source pad template as pad template, which
makes the link function think it can't link webmmux to shout2send.

Also add unit test.

https://bugzilla.gnome.org/show_bug.cgi?id=689336
 2012-11-30 17:22:34 +00:00
Wim Taymans
64cdbb77a9 rtspsrc: use new option parser function 2012-11-27 11:13:37 +01:00
Tim-Philipp Müller
5dee61a8d5 law: fix accidental file permissions change 
https://bugzilla.gnome.org/show_bug.cgi?id=687469
 2012-11-26 15:17:13 +00:00
Tim-Philipp Müller
314efb684b qtdemux: avoid criticals if unknown fourcc has space at beginning or end 
https://bugzilla.gnome.org/show_bug.cgi?id=682936
 2012-11-25 14:16:09 +00:00
Tim-Philipp Müller
efaa80fbc6 videobox: fix border filling for planar YUV formats 
We would get a green border instead of a black one, for
example.

https://bugzilla.gnome.org/show_bug.cgi?id=684991
 2012-11-24 19:32:51 +00:00
Tim-Philipp Müller
ef6c16a32e mulaw: const-ify some arrays 2012-11-24 14:27:33 +00:00
Roland Krikava
3be45f7022 mulawdec: fix integer overrun 
There might be more than 65535 samples in a chunk of data.

https://bugzilla.gnome.org/show_bug.cgi?id=687469
 2012-11-24 14:24:41 +00:00
Wim Taymans
5d0507c09e rtspsrc: pause the task instead of spinning 
Actually pause the loop task instead of spinning forever.
 2012-11-22 11:34:31 +01:00
Joshua M. Doe
fe9fb8d8a7 videoflip: Add gray 8/16 support 2012-11-20 12:49:49 +01:00
Wim Taymans
c28bfa8902 rtspsrc: handle segment event 
Make a segment event when we send a new range header to a client (first PLAY
request or after a seek). Send the segment event in interleaved mode.
Clean the segment event on cleanup

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688382
 2012-11-16 15:38:29 +01:00
Wim Taymans
bd91bd3193 rtspsrc: fix check for active streams 
A stream can be active without a srcpad yet and we want to send
events on those streams as well.
 2012-11-16 15:22:46 +01:00
Wim Taymans
11cf4d4fd3 rtspsrc: create and add pads outside of lock 
Create and add the ghostpad for the new stream outside of the lock because it
is not needed and causes deadlocks.
 2012-11-16 13:33:44 +01:00
Aleix Conchillo Flaque
6c855edf03 rtspsrc: allow client to disable reconnection 
* gst/rtsp/gstrtspsrc.[ch]: added new "udp-reconnect" property. Before,
  rtspsrc always tried to reconnect to the server when the RTSP
  connection was closed by the server. This property lets the user
  decide whether it wants rtspsrc to reconnect or not.

  https://bugzilla.gnome.org/show_bug.cgi?id=683912
 2012-11-16 12:55:10 +01:00
Wim Taymans
e2a4d28c1f rtspsrc: clear variables before retrying 
Else we might unref an old udpsrc twice in cleanup.
 2012-11-16 12:17:37 +01:00
Wim Taymans
cc9cb26be1 rtspsrc: propose ports in multicast 
When the user configured a port-range, propose ports from this range
as the multicast ports. The server is free to ignore this request but if it
honours it, increment our ports so that we suggest the next port pair for the
next stream.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=639420
 2012-11-16 12:17:37 +01:00
Wim Taymans
5025b3f1b3 rtspsrc: add more debug 2012-11-16 12:17:37 +01:00
Tim-Philipp Müller
6f1aa3e4d5 multifilesink: post messages in max-size mode as well 
No reason not to really.
 2012-11-16 09:13:22 +00:00
Wim Taymans
c33507f186 udpsrc: post error before stopping 2012-11-15 14:48:59 +01:00
Tim-Philipp Müller
bdf3c77828 gst_adapter_prev_timestamp -> gst_adapter_prev_pts 
https://bugzilla.gnome.org/show_bug.cgi?id=675598
 2012-11-14 00:13:36 +00:00
Nicolas Dufresne
673d2d24b8 videoflip: Add NV12/NV21 support 
https://bugzilla.gnome.org/show_bug.cgi?id=688225
 2012-11-13 14:25:04 +01:00
Wim Taymans
c755af0cb0 rtpsource: protect against invalid RTP packets 2012-11-12 11:18:30 +01:00
Tim-Philipp Müller
35fafae241 videocrop: add support for YV12 
We can do I420, so we can do YV12 as well.
 2012-11-10 18:21:28 +00:00
Alessandro Decina
b916d2b398 multifilesink: don't write stream headers with key-unit-event 
Don't write stream headers, let upstream elements insert them in the stream if
all_headers=true is set in key unit events.
 2012-11-10 12:41:33 +01:00
Nicolas Dufresne
e111068f7b videocrop: Add NV12/NV21 support 
https://bugzilla.gnome.org/show_bug.cgi?id=687964
 2012-11-10 01:52:44 +01:00
Sebastian Dröge
c70ba7765a udpsrc: Also clear GError 2012-11-09 11:22:30 +01:00
Sebastian Dröge
b86d20e45b udpsrc: Don't error out if we get an ICMP destination-unreachable message when trying to read packets 
See bug #529454 and #687782 and commit
751f2bb364
 2012-11-09 11:20:27 +01:00
Christian Fredrik Kalager Schaller
485505f323 Fix vp8rtp header names in Makefile 2012-11-07 13:36:33 +01:00
Nicolas Dufresne
1ad8ebac44 videocrop: Add support for automatic cropping 
This change enable automatic cropping using -1 set to left, top, right or
bottom property. In the case both side are set to automatic cropping, the
croping will be done equally on both side (in the odd case, right and
bottom cropping will be 1 pixel more).

https://bugzilla.gnome.org/show_bug.cgi?id=687761
 2012-11-07 11:20:24 +01:00
Marc Leeman
7cbca3dcd1 rtsp: the RTCP port number is inclusive 
The configured port number pair has its upper bound set to the maximum
allowed RTCP port, inclusive.

See https://bugzilla.gnome.org/show_bug.cgi?id=639420
 2012-11-06 13:22:58 +01:00
Tim-Philipp Müller
beb3c9c9be Fix FSF address 
https://bugzilla.gnome.org/show_bug.cgi?id=687520
 2012-11-04 00:09:59 +00:00
Tim-Philipp Müller
230cf41cc9 Fix FSF address 
https://bugzilla.gnome.org/show_bug.cgi?id=687520
 2012-11-04 00:07:18 +00:00
Wim Taymans
9857e6af4d vrawdepay: don't access rtp buffer after unmap 
Read the marker bit before we unmap the rtp packet.
 2012-11-02 18:48:17 +00:00
Douglas Bagnall
0b898ab911 videoconvert: Compare y offset with height, not width, when testing for overlap 
This could have prevented images showing that should have when the
source height is greater than its width.

When width exceeds height, as is common, it probably only caused a
miniscule amount of unnecessary work.  I haven't tested.
 2012-11-02 09:29:30 +01:00
Tim-Philipp Müller
5ac789408b rtpvp8: include config.h and minor style fixes 2012-11-01 21:10:21 +00:00
Tim-Philipp Müller
4a849d6690 rtp: fix tabs/space mess in Makefile.am 2012-11-01 20:53:48 +00:00
Tim-Philipp Müller
321acd14dc rtp: move VP8 payloader and depayloader from -bad 
Spec is still in draft state, but should hopefully not
change much now. Besides, we announce things as VP8-DRAFT-IETF-01
in our caps, so even if things change in incompatible ways it
should not break anything.

https://bugzilla.gnome.org/show_bug.cgi?id=687263
 2012-11-01 20:53:48 +00:00
Tim-Philipp Müller
44efab8e3d rtpvp8: use gst_element_class_set_static_metadata() 
where possible. Avoids some string copies. Also re-indent
some stuff. Also some indent fixes here and there.
 2012-11-01 20:53:48 +00:00
Mark Nauwelaerts
bc7dbbbd4f rtpvp8: replace gst_element_class_set_details_simple with gst_element_class_set_metadata 2012-11-01 20:53:48 +00:00
Sebastian Dröge
4853001547 rtpvp8: update for GST_PLUGIN_DEFINE() API changes 2012-11-01 20:53:48 +00:00
Wim Taymans
fccfca38d4 rtpvp8: update for buffer changes 2012-11-01 20:53:48 +00:00
Danilo Cesar Lemes de Paula
3edffb13e3 rtpvp8; fix compatibility with the third draft 
https://bugzilla.gnome.org/show_bug.cgi?id=671073
 2012-11-01 20:53:48 +00:00
Mark Nauwelaerts
d9581832a0 rtpvp8: port some more to new memory API 2012-11-01 20:53:47 +00:00
Olivier Crête
c6761daa27 rtpvp8: port to 0.11 2012-11-01 20:53:47 +00:00
Sebastian Dröge
2c5ea76bdc rtpvp8pay: Fix typo 2012-11-01 20:53:47 +00:00
Youness Alaoui
1cf155d70d rtpvp8: Update the pay/depay to the ietf-draft-01 spec 2012-11-01 20:53:47 +00:00
Vincent Penquerc'h
88aade4150 rtpvp8: fix bitstream parsing using the wrong kind of bitreader 
VP8 uses a probabilistic bool coder, not a straight bit coder.
This fixes parsing when error-resilient is set.

This commit includes a copy of libvpx's bool coder, BSD licensed.

https://bugzilla.gnome.org/show_bug.cgi?id=652694
 2012-11-01 20:53:47 +00:00
Olivier Crête
97c3f3617c rtpvp8: Reject unknown bitstream versions 2012-11-01 20:53:47 +00:00
Edward Hervey
74a1a704bf rtpvp8: Fix unitialized variable 
Makes macosx compiler happy.
 2012-11-01 20:53:47 +00:00
Sjoerd Simons
6ed6318076 rtpvp8depay: Accept packets with only one byte of data 
When fragmenting partions it can happen that an RTP packet only caries 1
byte of RTP data.
 2012-11-01 20:53:47 +00:00
Sjoerd Simons
a45e7a3fc0 rtpvp8pay: Treat the frame header just like any other partition 
When setting up the initial mapping just act as if the global frame
information is another partition. This saves special-casing it later in
the actual packetizing code.
 2012-11-01 20:53:47 +00:00
Sjoerd Simons
e9f4e9342f rtpvp8: Add simple payloaders and depayloaders for VP8 
Minimal implementation of http://www.webmproject.org/code/specs/rtp/,
version 0.3.2
 2012-11-01 20:53:47 +00:00
Wim Taymans
d6fd0ebd04 gstpay: fix for 1.0 events 
Caps events are sometimes not followed by a buffer but by an event. Flush any
pending caps before we make a packet with the event.
Chain up to the parent event handler before we attempt to push RTP packets, it
might be a segment event.
 2012-11-01 18:42:39 +00:00
Wim Taymans
05232c55a5 gstdepay: fix small leak 2012-11-01 18:42:24 +00:00
Wim Taymans
08e5a197b4 gstdepay: add support for events 
Conflicts:
	gst/rtp/gstrtpgstdepay.c
 2012-11-01 18:18:19 +00:00
Wim Taymans
54b783b5a3 rtpgstpay: add support for sending events 
We currently only send tags and custom events. The other events
might interfere with the receiver timings or are otherwise handled
by RTP.

Conflicts:
	gst/rtp/gstrtpgstpay.c
 2012-11-01 18:06:11 +00:00
Wim Taymans
6502d08e43 gstpay: rewrite payloader 
Use adapter to assemble the payload and make a flush function to
turn this payload into (fragmented) packets.

Conflicts:
	gst/rtp/gstrtpgstpay.c
	gst/rtp/gstrtpgstpay.h
 2012-11-01 17:57:52 +00:00
Douglas Bagnall
e3c77ba709 videomixer: get height via GST_VIDEO_FRAME_HEIGHT, not _WIDTH 
https://bugzilla.gnome.org/show_bug.cgi?id=687330
 2012-11-01 13:03:44 +00:00
Douglas Bagnall
79403bcb0c videbox: fix border filling for gray formats 
Get the height via GST_VIDEO_FRAME_HEIGHT, not _WIDTH.

https://bugzilla.gnome.org/show_bug.cgi?id=687330
 2012-11-01 13:02:16 +00:00
Wim Taymans
c0713e4b80 gstdepay: check for correct fragment offset 
Make sure we only insert the rtp packet in the adapter when the
frag_offset matches. When the first packet of a fragment is dropped,
it avoids putting the remaining packets in the adapter and processing
the partial fragment.

Conflicts:
	gst/rtp/gstrtpgstdepay.c
 2012-11-01 12:09:47 +00:00
Wim Taymans
8a402e0c06 gstpay: set C flag on all buffers of the fragment 
Set the C flags on all the fragments instead of only those with
caps in them. This makes it easier in the receiver to check if there
is a caps in the assembled fragments just by looking at the last RTP
packet flags.
 2012-11-01 12:06:08 +00:00
Wim Taymans
d78ff07f7d gstdepay: use the capsversion 
Take the caps from the input caps and store it in the slot given
by capsversion.
 2012-11-01 11:37:44 +00:00
Wim Taymans
936c3819b5 gstpay: send caps inline 
Place the capsversion on the outgoing caps so that they end up in
an SDP as well. Receivers need to know what capsversion a particular
caps is for to be able to match the caps to the CV in the RTP packets.
Place the caps inside the RTP packet whenever the caps change.

Based on patch by Andrzej Bieniek <andrzej.bieniek@pure.com>

Conflicts:
	gst/rtp/gstrtpgstpay.c
	gst/rtp/gstrtpgstpay.h
 2012-11-01 11:34:33 +00:00
Andrzej Bieniek
3b1931a039 gstpay: add debug 
Conflicts:
	gst/rtp/gstrtpgstpay.c
 2012-11-01 11:28:50 +00:00
Andrzej Bieniek
ee5ecc7773 depay: correctly skip caps header size 
Conflicts:
	gst/rtp/gstrtpgstdepay.c
 2012-11-01 11:27:13 +00:00
Tim-Philipp Müller
ef0805ea14 matroskademux: put streamheaders on vorbis/speex/flac/theora caps to make remuxing work 
https://bugzilla.gnome.org/show_bug.cgi?id=640589
 2012-10-30 23:29:46 +00:00
Tim-Philipp Müller
752cf98745 gst: fix variable order in some Makefile.am 
https://bugzilla.gnome.org/show_bug.cgi?id=687013
 2012-10-27 23:27:38 +01:00
Antoine Tremblay
a1c86de09a gst: add various missing GST_PLUGINS_BASE_LIBS in Makefile.am 
Those plugins depend on either libgstaudio or libgstvideo,
which are in gst-plugins-base.

https://bugzilla.gnome.org/show_bug.cgi?id=687013
 2012-10-27 23:26:41 +01:00
Alexey Fisher
29cd24bc41 matroskademux: mark invisible VP8 frames with the DECODE_ONLY flag 
https://bugzilla.gnome.org/show_bug.cgi?id=654259
 2012-10-27 14:46:02 +01:00
Stas Sergeev
238a5ec826 multifilesrc: fix stop index handling 
Make sure the stop index is always honoured. Avoids
endless loop if one wants to read and output the same
file N times, for example.

https://bugzilla.gnome.org/show_bug.cgi?id=654853
 2012-10-26 11:04:01 +01:00
Руслан Ижбулатов
78193dfe71 matroskademux: Support recursive SimpleTags 
Fixes #682644
Depends on #682615
 2012-10-26 10:16:42 +02:00
Руслан Ижбулатов
cd719bb808 matroskademux: Expand the tag mapping. 
* Also expose unknown tags as key=value pairs.
* Arrange tag map in the same order tags are listed in Matroska spec, leaving
unmapped tags as comments.
* More specific TODOs.
* Remove duplicate DATE define.

Fixes #682615
Depends on #682524
 2012-10-26 10:12:52 +02:00
Sebastian Dröge
6c635ce64f matroskademux: Fix uninitialized variable compiler warning 2012-10-26 10:09:39 +02:00
Руслан Ижбулатов
71fd688ef0 matroskademux: Matroska tag TargetType support 
* Reads TargetType and TargetTypeValue from a Tag.
* After Tag is completely read, processes taglist, substituting some of the
tags depending on target type value and the presence of video/subtitle streams.
* Supports reading two new simpletags - PART_NUMBER and TOTAL_PARTS

Depends on #682448
Fixes #682524
 2012-10-26 10:08:18 +02:00
Руслан Ижбулатов
b75628f041 matroskademux: Per-track tags for Matroska 
Requires Matroska file to have sane layout (track info before tag info).
Uses replace-merge.
Makes track UIDs 64-bit.

Fixes #682448
 2012-10-26 10:03:55 +02:00
Tim-Philipp Müller
fe7236230c multifilesrc: fix typo in property description 2012-10-25 20:19:44 +01:00
Michael Smith
a88caf84b4 qtdemux: read video format header fully (so we can find 'pasp' atoms) for more fourccs. 
Fixes aspect ratio of prores files.
 2012-10-25 12:18:50 -07:00
Thiago Santos
02d91dcd24 imagefreeze: the new get_caps already does the filter intersection 
It should be faster to pass the caps to intersect as the filter caps,
rather than using NULL and intersecting 'manually' later.

https://bugzilla.gnome.org/show_bug.cgi?id=686837
 2012-10-25 10:32:17 -03:00
Thiago Santos
115581eb2e imagefreeze: avoid assertion when using accept caps query 
This query must receive a fixed caps, so imagefreeze should
fixate its framerate before sending the query downstream.

https://bugzilla.gnome.org/show_bug.cgi?id=686837
 2012-10-25 09:39:36 -03:00
Arnaud Vrac
bc79fe565c qtdemux: use correct type for channel-mask bitmask 
Fixes crash on 32-bit systems.
 2012-10-24 12:54:08 +01:00
Tim-Philipp Müller
7275860bdd flacparse: fix coverart extraction if vorbis comments come after picture header 
See sample file for bug #684701.
 2012-10-23 16:02:05 +01:00
Tim-Philipp Müller
7c41f42eec flacparse: ignore bad headers if we have a valid STREAMINFO header 
If we run into any header parsing issues and we have a valid
STREAMINFO header already, don't error out, but just stop
header parsing and try to find some audio frames.

https://bugzilla.gnome.org/show_bug.cgi?id=684701
 2012-10-23 13:56:54 +01:00
Tim-Philipp Müller
49cc719809 flacparse: post proper error message and fix buffer leak on header parsing error 
https://bugzilla.gnome.org/show_bug.cgi?id=684701
 2012-10-23 13:56:54 +01:00
Michael Smith
150bd97e96 qtdemux: with raw audio, set a default channel-mask for multichannel audio. 
This doesn't actually parse 'chan' because it's absurdly complex.
 2012-10-22 22:34:43 -07:00
Sebastian Rasmussen
9fc62a58e3 updsrc: fix typo causing compilation error 
gstudpsrc.c: In function 'gst_udpsrc_create':
gstudpsrc.c:365: error: 'ret' may be used uninitialized in this function

https://bugzilla.gnome.org/show_bug.cgi?id=686642
 2012-10-22 23:19:28 +01:00
Wim Taymans
a2eead3d60 avi_ fix invert function 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686550
 2012-10-22 11:55:59 +02:00
Wim Taymans
0e3ef30c31 avi: fix debug 2012-10-22 11:55:22 +02:00
Wim Taymans
199aaa4021 qtdemux: add support for 'generic' samples 
Add support for stuffing a complete stream into 1 sample.

See https://bugzilla.gnome.org/show_bug.cgi?id=686550
 2012-10-22 11:39:37 +02:00
Tim-Philipp Müller
aa3ba65eb5 qtdemux: don't leak gst_riff_strf_auds in case of MS/RIFF audio 
https://bugzilla.gnome.org/show_bug.cgi?id=681192
 2012-10-19 19:26:45 +01:00
Mark Nauwelaerts
35cd53867c matroskamux: unsigned subtitle template 2012-10-19 16:14:01 +02:00
Youness Alaoui
13328bc129 videomixer2: Fix race condition where a src setcaps is ignored 
If both pads receive data at the same time, they will both get their
sink_setcaps called which will call the src_setcaps, but there is
a race condition where the second one might not be called.
Fixes: https://bugzilla.gnome.org/show_bug.cgi?id=683842
 2012-10-19 12:10:31 +02:00
Mark Nauwelaerts
5742352e10 matroskamux: do not use unoffical V_MJPEG codec id 
Since it's not spec'ed, consider it a VfW compatibility
case. Many applications (e.g. avidemux) don't understand
the unofficial V_MJPEG id.

Fixes #659837.

Conflicts:
	gst/matroska/matroska-mux.c
 2012-10-18 18:29:40 +01:00
Tim-Philipp Müller
488549bb54 Use gst_element_class_set_static_metadata() 
where possible. Avoids some string copies. Also re-indent
some stuff. Also some indent fixes here and there.
 2012-10-17 17:46:34 +01:00
Wim Taymans
e9040e90a5 jpegdepay: store quant tables in zigzag order 2012-10-17 14:23:01 +02:00
Wim Taymans
d5fd524a0c rtsession: fix compiler warning 2012-10-17 13:55:45 +02:00
Wim Taymans
26a21e85e2 rtpbin: clarify the ntp-sync option 2012-10-17 13:35:07 +02:00
Wim Taymans
f17db5c4ed rtpsession: update caps in the source 
Inform the source when caps changed. This was removed in the port to 1.0
leaving the source unaware of the clock-rate and unable to interpollate
rtp timestamps for SR packets.
 2012-10-17 13:22:40 +02:00
Wim Taymans
f4eef3f48d rtpbin: set PTS and DTS in jitterbufffer 2012-10-17 12:46:32 +02:00
Wim Taymans
796c1d8029 rtpbin: disable check for ntp-sync 
Disable the check for the ntp-sync method. It is expected that
a rather larger offset needs to be applied with this method.
 2012-10-17 12:27:03 +02:00
Wim Taymans
1cebcfa8c2 rtpbin: use running-time for NTP time 
When use-pipeline-clock is set, use the running-time of the
pipeline to calculate the NTP timestamps. This method would previously
only work when the base-time is set to 0 but with this change it can
also work with different offsets and we can also implement pause/resume
of the sender and receiver now.
 2012-10-17 12:26:05 +02:00
Wim Taymans
5ec642d0c3 videocrop: port to videofilter 2012-10-17 10:20:12 +02:00
Wim Taymans
3ef7c8ab93 videobox: use out_info for out properties 2012-10-17 09:36:50 +02:00
Wim Taymans
f701d980e6 median: small cleanups 2012-10-16 14:40:19 +02:00
Wim Taymans
0e21e80e9b median: remove now that it is in videofilter 2012-10-16 13:56:19 +02:00
Wim Taymans
9e67891f72 videomedian: copy media to videomedian 
Copy the median video filter to videofilters and rename to
videomedian.
 2012-10-16 13:47:24 +02:00
Wim Taymans
b893197317 media: port to 1.0 2012-10-16 13:16:29 +02:00
Tim-Philipp Müller
f94572fb36 avidemux: append palette data to paletted 8-bit RGB frames 
Fixes playback of 8-bit indexed RGB videos, with fixes in -base.

https://bugzilla.gnome.org/show_bug.cgi?id=686046
 2012-10-16 01:09:05 +01:00
Tim-Philipp Müller
e9682b938a qtdemux: don't assert if upstream size is not available when guessing bitrates 
Fixes abort in push mode where the source is not seekable and the
size of the file is not available, as with

  cat foo.mp4 | gst-launch-1.0 playbin uri=fd://0

Less noticable with releases, since we disable all
g_assert() there.

https://bugzilla.gnome.org/show_bug.cgi?id=686008
 2012-10-13 00:08:01 +01:00
Michael Smith
3a3a7c38aa qtdemux: allow more streams. Bump this constant to 32, which should be 
enough for real-world files.
 2012-10-12 14:38:33 -07:00