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1711 commits

Author SHA1 Message Date
Peter Kjellerstedt
db66ff4a62 rtsp: Rewrote setup_tunneling(). 
Rewrote setup_tunneling() to use normal GstRTSPMessages instead of hard
coded strings and duplicates of the message parsing code.
 2009-08-24 13:19:45 +02:00
Peter Kjellerstedt
c18e2eec88 rtsp: Rewrote gen_tunnel_reply(). 
Rewrote gen_tunnel_reply() to generate a normal GstRTSPMessage rather
than a hard coded string.
 2009-08-24 13:19:44 +02:00
Peter Kjellerstedt
e1b3393d6b rtsp: Ignore the Content-Length for POST requests. 
The Content-Length for POST requests with an x-sessioncookie header should
be ignored as the length is bogus and only there to fool proxies.
 2009-08-24 13:19:44 +02:00
Peter Kjellerstedt
11c8b811f3 rtsp: Normalize lines (remove extra whitespace) before parsing. 2009-08-24 13:19:44 +02:00
Peter Kjellerstedt
5716cd102a rtsp: Made parse_string() return a result. 
This will catch parsing errors when a too long string is received.
 2009-08-24 13:19:44 +02:00
Peter Kjellerstedt
fdd5a65632 rtsp: Improved parsing of messages. 
Do not abort message parsing as soon as there is an error. Instead parse
as much as possible to allow a server to return as meaningful an error as
possible.
 2009-08-24 13:19:44 +02:00
Peter Kjellerstedt
ca154010fe rtsp: Added support for HTTP messages 2009-08-24 13:19:44 +02:00
Peter Kjellerstedt
dd7d0cfc45 rtsp: Added gst_rtsp_connection_create_from_fd(). 
API: gst_rtsp_connection_create_from_fd()
 2009-08-24 13:19:44 +02:00
Peter Kjellerstedt
814eaa728a rtsp: Add initial buffer support. 
The initial buffer contains data for a connection which should be used
before starting to actually read anything from the socket.
 2009-08-24 13:19:44 +02:00
Wim Taymans
2c08c76383 appsink: don't block in paused 
When we are asked to unlock we should either leave the render function or call
the wait_preroll method to release the stream lock.

Fixes #592657
 2009-08-24 13:16:39 +02:00
Peter Kjellerstedt
41f1d9a7d9 rtsp: Add support for the Authentication-Info header. 
The Authentication-Info header is defined in RFC 2617 (Digest Access
Authentication).
 2009-08-24 11:24:27 +02:00
Peter Kjellerstedt
3c4fa9274f rtsp: Avoid duplicated headers. 
Remove any existing Session and Date headers before adding new ones
when sending a request. This may happen if the user of this code reuses
a request (rtspsrc does this when resending after authorization fails).
 2009-08-19 09:31:51 +02:00
Peter Kjellerstedt
3b888cfe2a rtsp: Corrected the HTTP digest authorization computation. 
Do not use sizeof() on an array passed as an argument to a function and
expect to get anything but the size of a pointer. As a result only the
first 4 (or 8) bytes of the response buffer were initialized to 0 in
auth_digest_compute_response() which caused it to return a string which
was not NUL-terminated...
 2009-08-18 16:50:58 +02:00
Mark Nauwelaerts
87e6775844 riff: align API doc of gst_riff_parse_chunk with reality 2009-08-12 13:39:14 +02:00
Tim-Philipp Müller
cb19626c8c rtspconnection: don't use GLib-2.18 function 
g_checksum_reset() was added only in GLib 2.18, but we still require
only 2.16, so work around that if we only have 2.16. Fixes #591357.
 2009-08-10 20:18:24 +01:00
Sebastian Dröge
79ade6ad68 rtsp: Use GLib's GChecksum instead of our own MD5 implementation 2009-08-10 10:19:01 +02:00
Mart Raudsepp
689a4d4c10 navigation: Fix doc blurb typo for gst_navigation_send_key_event 2009-08-09 20:52:40 -04:00
Tim-Philipp Müller
0021e6b765 Revert inlines that cause compiler warnings and are not needed anyway 2009-08-08 17:51:10 +01:00
Edward Hervey
9329b8be72 gst-libs: Remove dead assignments and resulting unused variables. 2009-08-08 15:54:57 +02:00
Wim Taymans
090808a295 baseaudiosrc: change default slave method 
Set the default slave method to the much better skew slaving algortihm.
 2009-08-06 12:58:58 +02:00
John Millikin
cd31b2e298 tag: Add support for ALBUM_ARTIST tag in vorbiscomments and ID3v2 tags 
Require latest core for this.

Fixes bug #590430.
 2009-08-06 06:43:38 +02:00
Sebastian Dröge
713f6ca8d5 cddabasesrc: Allow to specify the device name in the URI 
The allowed URI scheme is now:
cdda://(device#)?track

Also allow every combination of uppercase and lowercase
characters for the protocol part.

Fixes bug #321532.
 2009-08-06 06:43:34 +02:00
Philip Jägenstedt
1b4220bd03 appsrc: Clarify documentation about caps and linkage 
Fixes bug #589095.
 2009-08-06 06:43:34 +02:00
Olivier Crête
429d3555a2 audiofilter: Don't assert on slightly different caps 
Plugins should not assert on incompatible caps, caps negotiation will
fail anyway.
 2009-07-30 14:34:05 +01:00
Olivier Crête
4e88633de4 audiosink: Add stream-status messages 
Fixes #587695
 2009-07-20 12:54:37 +02:00
Olivier Crête
cc0da016f8 audiosrc: Add stream-status messages 
See #587695
 2009-07-20 12:54:37 +02:00
Tim-Philipp Müller
d53e754d42 typefinding: use subtitle/x-kate for Kate subtitle streams and application/x-kate for the rest 
Differentiate subtitle streams and lyrics/cracktastic/complex streams via
the category string in the headers. This seems like a useful distinction
to make, and also seems more future-proof. See #525743.
 2009-07-13 23:00:04 +01:00
Stefan Kost
cae6a55ba3 navigation: simplify docs 
Make short-desc short - its used in the toc. Strip uneeded markup.
 2009-07-13 21:54:47 +03:00
Jan Schmidt
85de44aa01 navigation: Add some partial documentation 
Add a general documentation blurb for the GstNavigation functionality.
Still lacks some example code and detail on how to implement it.
 2009-07-13 17:55:55 +01:00
Tim-Philipp Müller
f6a508d963 pbutils: add description for Siren codec and make two descriptions non-translatable 2009-07-13 17:52:39 +01:00
Elliott Sales de Andrade
132fb5c050 riff: add siren to the RIFF parser 
Add siren7 caps to the RIFF parser.
 2009-07-13 18:22:55 +02:00
David Schleef
530cb7268b basevideo: send basevideo back to remedial school 
Move basevideo classes and schroedinger plugin to -bad.
 2009-07-01 10:27:30 -07:00
Wim Taymans
6c28c3f139 netaddress: add constant for max len 2009-07-01 12:54:21 +02:00
Wim Taymans
8ef62de3f0 netbuffer: add gst_netaddress_to_string 
Add function to serialize a net address to a string.

API: GstNetAddress::gst_netaddress_to_string()
 2009-07-01 12:48:38 +02:00
Stefan Kost
0e967f1b14 multichannel: rewrite the new doc comment a bit 
Its part of the audio lib.
 2009-06-29 17:49:58 +03:00
Wim Taymans
8601862e27 ringbuffer: add vmethod to clear the ringbuffer 
Add a vmethod so that subclasses can be notified when they should clear the data
in the ringbuffer.
 2009-06-29 15:17:25 +02:00
Jan Schmidt
a9097080a3 riff-media: Fix the fourcc caps property for VC-1/WMVA 
The caps property for carrying fourccs is 'format', not 'fourcc'
 2009-06-29 14:01:33 +01:00
Wim Taymans
f5962f0a4f rtsp: include in.h for FreeBSD compat 
Fixes #586920
 2009-06-29 12:20:52 +02:00
Wim Taymans
3928dbbb45 appsink: add docs and signals 
Add docs for the new callback.
Add signals for the new buffer-list support.
 2009-06-29 12:14:43 +02:00
Branko Subasic
6518d283d5 Added buffer list support. 2009-06-29 11:59:47 +02:00
Branko Subasic
fb0fd53212 Added buffer list support. 2009-06-29 11:59:46 +02:00
Peter Kjellerstedt
8927dbc98b sdp: Include winsock2.h after defining WINVER. 
Similar to bug #587080.
 2009-06-29 09:36:27 +02:00
Peter Kjellerstedt
c398f2f376 rtsp: Moved a comment. 2009-06-29 09:31:40 +02:00
Stefan Kost
57a7d6f699 docs: add basic section docs for multichannel and relocate the ones for audio 
Add section docs for multichannel, so that it has a short desc in the toc too.
Move the section docs in adio up, so that the follow the copyright like
elsewhere.
 2009-06-27 23:25:09 +03:00
Руслан Ижбулатов
07c237ad19 Define WINVER before including any win headers 
Fixes bug #587080.
 2009-06-27 14:02:50 +02:00
René Stadler
41b7504e9c riff: prevent crash if rounded up tag size exceeds data size 
When rounding up `tsize' exceeds the remaining buffer size, `size' underflows
and an invalid read past the buffer data follows.
 2009-06-27 01:22:52 +03:00
Sebastian Dröge
939baee2bd basevideocodec: By default don't allow caps changes on the srcpad 
This fixed playback of Dirac files with schrodec when upstream wants
a different width/height, basevideocodec accepts this and then
pushes buffers with new caps but content of the old caps.
In the best case this will just result in wrong unit size and a
failure in basestransform elements.
 2009-06-26 15:20:09 +02:00
Tim-Philipp Müller
adff66fc83 pbutils: add description for multipart 
So we get slightly nicer error messages when multipartdemux is missing.
 2009-06-24 09:51:11 +01:00
Wim Taymans
85af9b82e8 basertppayload: add support for bufferlists 
Based on patch from Ognyan Tonchev.

See #585559
 2009-06-19 15:52:34 +02:00
Wim Taymans
f5c8055edf rtpbuffer: use new convenience functions 
New core convenience functions makes the list getters and setters trivial.
Maybe even too trivial...
 2009-06-19 15:33:04 +02:00
Wim Taymans
457d39075c rtp: cleanups, add _list_get_seq() too 
Clean up the docs a little.
Add missing _list_get_seq method.
Add new symbols to the docs
 2009-06-18 19:04:52 +02:00
Wim Taymans
e2ccc1ee39 rtp: cleanups 
Add Since tags to docs
Move some code around
Add win32 symbols
 2009-06-18 18:51:04 +02:00
Wim Taymans
66c388a0e0 rtp: add bufferlist support 2009-06-18 18:51:04 +02:00
Wim Taymans
f385081c92 rtp: pass data to macros instead of GstBuffer 2009-06-18 18:50:35 +02:00
Peter Kjellerstedt
4fd61fbaa4 rtsp: Made the parsing of the RTSP URL scheme more generic. 2009-06-17 18:34:57 +02:00
Peter Kjellerstedt
726a47f777 rtsp: Added gst_rtsp_watch_queue_data(). 
gst_rtsp_watch_queue_data() is similar to gst_rtsp_watch_queue_message()
but allows for queuing any data block for writing (much like
gst_rtsp_connection_write() vs. gst_rtsp_connection_send().)

API: gst_rtsp_watch_queue_data()
 2009-06-17 18:34:33 +02:00
Peter Kjellerstedt
595f8b6d00 rtsp: Only extract the session ID from RTSP responses. 2009-06-17 18:02:18 +02:00
Peter Kjellerstedt
ddbeb44f14 rtsp: Added support for parsing IPv6 addresses in RTSP URLs. 2009-06-17 18:00:17 +02:00
Peter Kjellerstedt
95a606a0bb rtsp: Use getaddrinfo() to support both IPv4 and IPv6. 2009-06-17 17:59:47 +02:00
Peter Kjellerstedt
e1a4c8871a rtsp: Improved base64 decoding in fill_bytes(). 
The base64 decoding in fill_bytes() expected the size of the read data to
be evenly divisible by four (which is true for the base64 encoded data
itself). This did not, however, take whitespace (especially line breaks)
into account and would fail the decoding if any whitespace was present.
 2009-06-17 17:53:54 +02:00
Wim Taymans
ffd90dda89 audiosrc: fix get_offset 
When we need to jump to the most recently captured sample, jump to where the
next sample will be written instead of to some old data.

Fixes #581460
 2009-06-17 14:00:23 +02:00
Wim Taymans
57a13f28de audiosink: free the ringbuffer when going to NULL 
Unparent and free the ringbuffer when going to NULL, like we do with the
audiosrc element. We can do this now because we correctly manage the time
jumping back to 0.
 2009-06-17 13:18:18 +02:00
Wim Taymans
e4492c24ea audio: correctly handle short read/writes 2009-06-17 13:17:30 +02:00
René Stadler
2c5f455423 baseaudiosrc: add some extra logging for buffer timestamps 2009-06-17 12:36:50 +02:00
Sebastian Dröge
a64caea0bd videofilter: Add a default get_unit_size function 
This returns the correct values for all formats that are handled by
GstVideoFormat and makes all the custom get_unit_size functions in
many elements unnecessary.
 2009-06-16 19:38:17 +02:00
Wim Taymans
33837d420c rtsp: add Timestamp header field 
fixes #585994
 2009-06-16 18:57:20 +02:00
Tim-Philipp Müller
70089160f8 audiosink, audiosrc: do the class_ref()s in the right class_init functions 
Spotted by Philip Jägenstedt. Hopefully fixes #585970 for real.
 2009-06-16 14:14:26 +01:00
Tim-Philipp Müller
3767cb6005 audiosink,audiosrc: ref the audio ring buffer class and type in class_init 
Hack around thread-safety issues in GObject and our racy _get_type()
functions (we could easily fix the _get_type() functions, but we still
need to hack around the GObject class races until we require a newer
GLib version, I think).
 2009-06-15 15:39:09 +01:00
Wim Taymans
a5491ba218 audiosrc: return FALSE when receiving a SEEK event 
When receiving a seek event, return FALSE as we don't implement seeking.
 2009-06-15 12:57:39 +02:00
Peter Kjellerstedt
73dd8236ce rtsp: Use a more consistent naming of GstRTSPRec variables. 2009-06-15 09:28:34 +02:00
Peter Kjellerstedt
ff38999c8b rtsp: Call message_sent() callback for all sent messages. 
Previously the messages_sent() callback was only called for messages
which had a CSeq, which excluded all data messages. Instead of using the
CSeq as ID, use a simple index counter.
 2009-06-15 09:28:13 +02:00
Wim Taymans
a9c82f9472 ringbuffer: handle border cases in resampler 2009-06-11 19:13:28 +02:00
Wim Taymans
8bbf2e8a32 docs: fix typo 2009-06-11 12:39:19 +02:00
Wim Taymans
69b7fb3845 baseaudiosink: reset accum when dropping samples 
When we are resampling and we drop samples because we paused, reset the accum
counter because it's now invalid.
 2009-06-11 12:38:35 +02:00
Jan Schmidt
c1bc55a4f5 docs: Fix a couple of warnings from the docs build. 2009-06-11 11:16:15 +01:00
Tim-Philipp Müller
249d9b4aa1 Don't include config.h multiple times when build audio testchannel app. 
Fixes build problem on win32 (#585075).
 2009-06-10 21:37:29 +01:00
Wim Taymans
e01fab3ace rtsp: add some more docs 2009-06-09 22:00:53 +02:00
Peter Kjellerstedt
263c5b227b rtsp: Avoid a compiler warning. 2009-06-09 18:24:55 +02:00
Peter Kjellerstedt
dfc57e3f8a rtsp: Updated documentation for GstRTSPResult. 
Moved GST_RTSP_ELAST to be last in the documentation to match the actual
enum values.
 2009-06-09 18:23:28 +02:00
Peter Kjellerstedt
9c40eeeb4c rtsp: Plug a memory leak. 
Free memory related to any partially read and/or written RTSP messages.
 2009-06-09 16:28:20 +02:00
Wim Taymans
38e59ec75d baseaudiosink: no need to cause discont when clipping 
Remove the discont-when-clipping hack now that basesink provides us with
correctly clipped samples when stepping.
 2009-06-09 12:09:15 +02:00
Wim Taymans
cb4952fc2e audiosink: don't align when we clip 
Don't align samples when they were clipped. Not entirely correct but better than
nothing for now.
 2009-06-08 17:26:59 +02:00
Edward Hervey
ee3b251234 pbutils: Add description for hdv/aux-* formats. 2009-06-08 10:25:00 +02:00
Tim-Philipp Müller
5da78c8489 libgsttag: don't extract genres from empty ID3v1 tags 
If we don't have any other info, don't try to interpret the
genre field. In particular we don't want to interpret a genre
of 0 as 'Blues' if no other fields are set and the entire tag
is just empty.
 2009-06-06 12:04:12 +01:00
Peter Kjellerstedt
2dbd8702dd rtsp: Fixed a typo. 2009-06-05 14:06:17 +02:00
Peter Kjellerstedt
de18ad458f rtsp: Remove an unused variable. 2009-06-05 14:05:54 +02:00
Peter Kjellerstedt
b0a9848524 rtsp: Removed duplicate initialization of conn->writefd. 2009-06-05 13:59:14 +02:00
Peter Kjellerstedt
0167e3589d rtsp: Use #defined status codes. 2009-06-05 13:55:08 +02:00
Peter Kjellerstedt
c1a6644a18 rtsp: Correct gen_tunnel_reply(). 
Prevent gen_tunnel_reply() from generating an incomplete response
in case an error response code is given.
 2009-06-05 13:53:29 +02:00
Wim Taymans
59d9833924 rtsp: add G_LIKELY because we can 2009-06-02 12:10:39 +02:00
Peter Kjellerstedt
d8e0b5a4da rtsp: Avoid compiler warnings with -Wextra. 2009-06-01 09:59:22 +02:00
Peter Kjellerstedt
848b834cb9 rtsp: Include gst/gstconfig.h to make sure GST_PADDING is defined. 2009-06-01 09:58:27 +02:00
Peter Kjellerstedt
e69c3a4f70 sdp: Remove an unused variable. 2009-06-01 09:43:04 +02:00
Wim Taymans
dcc42d5f92 netbuffer: also note the order of IP4 addresses 
IP4 addresses are also stored in network byte order. Make a note of this in the
docs.
 2009-05-27 11:08:37 +02:00
Tim-Philipp Müller
6292ff4ae0 Revert "rtspconnection: don't use GLib-2.16 API, we require only 2.14" 
This reverts commit 418760cf74.

We now require GLib 2.16.
 2009-05-26 18:21:31 +01:00
Wim Taymans
796f8e2f76 netbuffer: document that the port is network order 
Document the fact that we store the port number in network order in
GstNetAddress and that the caller should byteswap appropriately.
 2009-05-26 15:39:18 +02:00
Andy Wingo
c7ca6abe53 add can-activate-pull property to baseaudiosink 
* gst-libs/gst/audio/gstbaseaudiosink.c: Add can-activate-pull property
  to baseaudiosink.
 2009-05-26 13:17:44 +02:00
Bastien Nocera
9c508ba458 cddabasesrc: Remove copy of sha1 digest 
Remove our copy of sha1 digest now that we depend on glib 2.16.
Fixes #536313
 2009-05-26 11:11:03 +02:00
Tim-Philipp Müller
5fa9a8f4d0 video: don't expose internal gst_adapter_get_buffer() helper function 
If it's really needed it should go into GstAdapter in core.
 2009-05-25 00:19:25 +01:00
David Schleef
538c1cde31 basevideo: Fix memleak 2009-05-22 21:29:51 -07:00
David Schleef
35aae561e8 basevideo: Add preset interface to encoder 2009-05-22 17:34:56 -07:00
Wim Taymans
81170c4989 audiosink: improve debug message 2009-05-21 10:48:49 +02:00
Michael Smith
35a9de28f4 gstid3tag: Don't extract a track number unless present. 
In ID3v1, a track number is present only if byte 125 is null AND
byte 126 is non-null. If the track number is not present, don't add
a track number tag with value 0.
 2009-05-19 18:12:18 -07:00
Wim Taymans
243d366b34 videoutils: remove adapter methods 
Remove adapter methods now that they are in core.
 2009-05-20 00:48:40 +02:00
Wim Taymans
c68a361e31 audiosink: return the return value of wait_preroll 
Return the value that _wait_preroll() returned instead of always WRONG_STATE.
 2009-05-19 17:17:37 +02:00
David Schleef
17f3810f7b video: remove // comments 2009-05-15 16:21:15 -07:00
David Schleef
45cf881f39 video: Add Y444, v210, v216 formats 2009-05-15 16:18:59 -07:00
David Schleef
4ec34e83d5 video: Copy BaseVideo classes from Schroedinger 2009-05-15 16:18:58 -07:00
Tim-Philipp Müller
f2031e1313 pbutils: add descriptions for 3GP, JPEG 2000 and Motion JPEG 2000 2009-05-15 20:50:06 +01:00
Wim Taymans
b9723f6e1c audioclock: make our internal time monotonic 
Make the internal time increase monotonically.
 2009-05-13 21:38:56 +02:00
Sebastian Dröge
ab75db1653 propertyprobe: Fix typo in the docs 2009-05-12 15:53:07 +02:00
Wim Taymans
0a09632396 rtpdepay: add some more comments 2009-05-12 10:39:49 +02:00
Wim Taymans
d655120ee6 audioclock: make sure values are ever increasing 2009-05-12 10:39:41 +02:00
Sebastian Dröge
24dd91b1f0 interfaces: Seperate some more struct definitions from typedefs 2009-05-12 09:03:25 +02:00
Sebastian Dröge
e057414049 interfaces: Seperate some more struct definitions from typedefs 2009-05-12 09:03:25 +02:00
Sebastian Dröge
59aa1251d9 interfaces: API: Add gst_mixer_get_mixer_type() 
This is a convenience function that returns the mixer_type
of the interface struct.
 2009-05-12 09:03:24 +02:00
Sebastian Dröge
29b063b39b interfaces: Add docs for gst_color_balance_get_balance_type() 2009-05-12 09:03:24 +02:00
Sebastian Dröge
9fc4d195e1 vorbistag: Use text/uri-list as mimetype instead of ---> for URI lists 2009-05-12 09:03:22 +02:00
John Millikin
ef473dd0ae vorbistag: Store cover art in vorbiscomments 
Fixes bug #513373.
 2009-05-12 09:03:22 +02:00
Sebastian Dröge
e1875bf25f interfaces: API: Add gst_color_balance_get_balance_type() 
This is a convenience function that returns the balance_type
of the interface struct.
 2009-05-12 09:03:22 +02:00
Sebastian Dröge
b6c3567b41 interfaces: Separate struct definitions from typedefs 2009-05-12 09:03:22 +02:00
Tim-Philipp Müller
279b996d20 pbutils: add description for APE tag caps 2009-05-12 01:59:01 +01:00
Tim-Philipp Müller
3d33e2a873 tagdemux: cache events from upstream and re-send them once we have a source pad 
Makes sure tags don't get dropped when we have multiple tag demuxers in a row.
Fixes #580318.
 2009-05-12 01:15:21 +01:00
Michael Smith
8f6399f109 riff: support UYVY raw 4:2:2 in riff. 2009-05-11 14:04:16 -07:00
Andy Wingo
9f74ce745f Revert "add can-activate-pull property to baseaudiosink" 
This reverts commit c4074a2ee4.
 2009-04-29 11:18:42 +02:00
Andy Wingo
219a31fa3c Revert "[baseaudiosink] add docs for can-activate-pull" 
This reverts commit 416ce16f26.
 2009-04-29 11:18:33 +02:00
Andy Wingo
416ce16f26 [baseaudiosink] add docs for can-activate-pull 
* gst-libs/gst/audio/gstbaseaudiosink.c: Add documentation for
  can-activate-pull.
 2009-04-28 18:48:33 +02:00
Andy Wingo
c4074a2ee4 add can-activate-pull property to baseaudiosink 
* gst-libs/gst/audio/gstbaseaudiosink.c: Add can-activate-pull property
  to baseaudiosink.
 2009-04-28 18:28:50 +02:00
Tim-Philipp Müller
8efe6108c4 cddabasesrc: fix posting of discid tags after MERGE_MODE_REPLACE_ALL changes in core 
Don't use REPLACE_ALL merge mode when that's not really what we want,
as now that REPLACE_ALL actually does what it's supposed to do in
core, we drop tags we wanted to keep, such as the various disc id
tags. Add unit test for this as well. Fixes #579463.
 2009-04-19 18:15:28 +01:00
Tim-Philipp Müller
418760cf74 rtspconnection: don't use GLib-2.16 API, we require only 2.14 
Fixes #579267.
 2009-04-17 10:35:34 +01:00
Wim Taymans
32904de58f baseaudiosink: don't unparent the ringbuffer 
when going to NULL, don't unparent the ringbuffer because we don't support going
back to 0 very well yet.
Fixes #579203
 2009-04-17 11:03:32 +02:00
Olivier Crete
d927114ef8 RTCP: don't fail when retrieving invalid PT 
We can't meaningfully assert on valid packet types so just return the type as it
is. Update the comments to reflect this.

Fixes #579192.
 2009-04-17 10:53:10 +02:00
Wim Taymans
f83f57b648 app: add trivial cast macros 
Add trivial cast macros for appsrc and appsink. Mark them as being since 0.10.23
and add the macros to the standard macros in the docs.

Fixes #579130
 2009-04-16 12:14:43 +02:00
Sebastian Dröge
a6cf0c8f06 video: Fix typo in the docs 2009-04-15 15:35:59 +02:00
Sebastian Dröge
a1d8cfde9d video: Add support for YVYU YUV colorspace 2009-04-15 14:53:47 +02:00
Tim-Philipp Müller
75acca2835 docs: fix hyperlink and move fft attribution to the right place 2009-04-15 00:19:19 +01:00
Stefan Kost
ab24d9d65c log: use G_GUINT64_FORMAT instead of llu 2009-04-15 00:02:39 +03:00
Josep Torra
71ab187355 RTSP: add missing headers for WMS RTSP 
Add missing headers related to Windows Media RTSP extension.
Fixes #578942
 2009-04-14 18:31:52 +02:00
Tim-Philipp Müller
9f23b82b2c Give credit to Mark Borgerding (kissfft author) 
and add myself to AUTHORS as well. Fixes #575638.
 2009-04-14 17:11:19 +01:00
Johann Prieur
86edcadc43 RTCP: add beginnings of Feedback messages 
Add the beginnings of parsing and constructing Feedback messages.
Fixes #577610.
 2009-04-14 16:45:20 +02:00
Wim Taymans
dffd1bcc97 baseaudiosrc: adjust the internal timestamp 
Adjust the internal timestamp before comparing it against the adjusted clock
time.
Fixes #578506
 2009-04-14 13:16:14 +02:00
Wim Taymans
0c4c1410f9 baseaudiosink: use new clock time methods 
Use the unadjusted internal clock times to calculate the internal/external
offset when calibrating the clock.

When going to NULL, unparent and free the ringbuffer, like we do in the source
element.
See #578506
 2009-04-14 13:12:59 +02:00
Wim Taymans
4231d54823 audioclock: add methods for the internal offset 
Add two methods for getting the unadjusted time of the clock and one for
adjusting an internal time. We will need these methods for correctly handling
the time after a gst_audio_clock_reset().

Add a debug category and some debug lines to the audio clock.

API: gst_audio_clock_get_time()
API: gst_audio_clock_adjust()
API: GST_AUDIO_CLOCK_CAST()
 2009-04-14 13:08:52 +02:00
Wim Taymans
251f152c20 baseaudiosink: use the internal clock time 
We can't assume that the internal clock time is the same as the function we
installed on our provided clock because somebody might have changed it.
 2009-04-10 21:50:55 +02:00
Martin Samuelsson
ee03bf5379 appsink: make callbacks return GstFlowReturn 
Make the new_buffer and new_preroll callbacks return a GstFlowReturn so that
errors can be reported properly.
Fixes #577827.
 2009-04-09 23:46:17 +02:00
Wim Taymans
e6798c5cce ringbuffer: allow for custom commit functions 
Allow subclasses to override the commit method.
 2009-04-09 18:04:44 +02:00
Wim Taymans
cae2981f83 baseaudiosink: fix a small glitch after pause 
After we pause the stream and interrupt the writeout to the ringbuffer, also adjust
the amount of output samples we consumed. We can't do this reliably with the
current API when we are doing trick modes but we can do the right thing for
normal playback.
 2009-04-08 18:06:54 +02:00
Stefan Kost
ff9ee1dc5a audiofilter: don't leak pad-template 
gst_element_class_add_pad_template() does not take ownership.
 2009-04-07 22:39:07 +03:00
Edward Hervey
2555eeb737 navigation/v4l: Don't use g_return_val_if_fail for computed/used values. 2009-04-04 16:28:14 +02:00
Wim Taymans
88110ea67e rtsp: use fully qualified urls when using a proxy 
Use a fully qualified url when specifying the url for tunneled requests through
a proxy.
See #573173
 2009-04-02 22:28:55 +02:00
Jan Schmidt
033e654172 navigation: Extend the navigation interface 
Add support for a set of standard commands that can be queried and executed to
support applications like DVD. Add query construction and parsing functions.
Add new messages that can be sent on the bus to provide notifications related
to commands, multiangle changes, and button highlight activity.
Add some helper functions to parse the existing GstNavigation events that
elements might receive.
Document it all and add unit tests.
 2009-04-02 12:21:18 +01:00
Wim Taymans
eed784b372 rtsp: fix little typo in the comments 2009-04-01 09:03:35 +02:00
Tim-Philipp Müller
fc8c5cba15 rtspconnection: make gst_rtsp_watch_queue_message() thread-safe 
People might queue messages from a thread other than the thread in which
the main context which this watch is attached is iterated from, so use
a GAsyncQueue instead of a GList, so g_list_append() doesn't trample
over list nodes just freed in the other thread. This just fixes issues
I've had with gst-rtsp-server. We might need more locking in various
places here.
 2009-03-31 18:30:57 +01:00
Tim-Philipp Müller
dfe96ce618 rtsp: clear the entire builder structure 
And use structure instead of variable with sizeof when
clearing the rtsp message structure, for clarity.
 2009-03-31 18:30:48 +01:00
Tim-Philipp Müller
dd9f077177 docs: fix typo in gst_rtsp_message_unset() API docs 2009-03-31 18:30:48 +01:00
Wim Taymans
8b37dc3eb8 rtsp: add support for proxies 
Add suport for proxy servers. Currently only used for tunneled HTTP
connections without authentication.
 2009-03-31 19:00:00 +02:00
Wim Taymans
8be68f983c Revert "rtsp: reset whole message (was sizeof pointer instead of sizeof type)" 
This reverts commit 79de0b8d67.
 2009-03-31 18:57:08 +02:00
Stefan Kost
79de0b8d67 rtsp: reset whole message (was sizeof pointer instead of sizeof type) 2009-03-31 12:27:09 +03:00
Jan Schmidt
43788e4796 doc: Fix a typo in the GstMixer docs 2009-03-31 00:58:24 +01:00
Wim Taymans
0d3d3026d2 rtsp: start CSeq counting from 1 instead of 0 
Start counting from 1 instead of 0 as this is what most other clients
seem to do.
 2009-03-25 16:37:28 +01:00
Wim Taymans
1081ae59eb rtsp: add ETag and If-Match headers 
Add new headers, we need them for RealMedia support.
 2009-03-25 16:36:14 +01:00
Tim-Philipp Müller
0267e79778 audiosrc: improve 'Dropped n samples' warning message 2009-03-25 11:27:44 +00:00
Sebastian Dröge
108ead73c8 rtsp: Use GLib base64 functions and deprecate gst_rtsp_base64_encode 
This also fixes another instance of CVE-2008-4316.
 2009-03-17 22:53:44 +01:00
Wim Taymans
f4b7cbbf16 rtsp: fix resolving of hostnames 
We were returning a pointer to a stack variable with the resolved hostname,
which doesn't work.
return a copy of the resolved ip address instead.
Fixes #575256.
 2009-03-13 16:19:41 +01:00
Wim Taymans
91b2d71da0 appsrc: release lock in _eos flushing case 
Release the mutex when we are flushing in gst_app_src_end_of_stream()
Fixes #574964.
 2009-03-13 15:16:44 +01:00
Jan Schmidt
566583e871 vorbistag: Protect memory allocation calculation from overflow. 
Patch by: Tomas Hoger <thoger@redhat.com> Fixes CVE-2009-0586
 2009-03-12 15:02:07 +00:00
Wim Taymans
0e2157029e rtsp: fix parsing of the timeout parameter 
--
 2009-03-11 18:45:59 +01:00
Wim Taymans
b674584e97 rtsp: fix g_return condition 
when parsing a data message, we require a data message.
 2009-03-11 17:29:41 +01:00
Wim Taymans
18f612ffa9 rtsp: free the right string. 
Free the key value before we remove the header item from the array. The item we
retrieved from the array is only valid until we remove it from the array.
 2009-03-11 14:09:54 +01:00
Wim Taymans
16225d45be rtsp: keep track of amount of decoded bytes 
Keep track of the actual amount of decoded bytes, which can be less than 3 when
we decode the last bits of a base64 message.
 2009-03-11 14:09:54 +01:00
Wim Taymans
f964c0fc38 rtsp: only add ports when not using TCP 
Only add the port numbers in the transport string when we are using udp or
multicast.
 2009-03-09 13:53:41 +01:00
Wim Taymans
bc54a5f9a0 rtsp: use gstreamer dump mem 
--
 2009-03-09 13:53:15 +01:00
Wim Taymans
3a72044a22 rtsp: use glib base64 encoder 
--
 2009-03-09 13:51:48 +01:00
Edward Hervey
a3c88fb32b Riff: Add mapping for Fraps video codec. 
Found through insanity testrun. Confirmed mapping in libavformat.
 2009-03-09 10:03:13 +01:00
Edward Hervey
b870b61c00 riff: Add the 'DVR ' mapping for mpeg2video. 
Found this in 3 files from the insanity suite and mapping is also present
in libavformat.
 2009-03-09 09:08:00 +01:00
LRN
eb3ff95a3a rtsp: fix compilation on windows. 
Remove unused variable when building for windows.
Fixes #574443.
 2009-03-08 18:17:48 +01:00
Wim Taymans
d998f6097b riff: add theora mapping 
Add theora mappings. See #574169.
 2009-03-06 18:54:57 +01:00
Wim Taymans
2cc1a6808d rtsp: Add methods for getting the read/write fds 
API:gst_rtsp_connection_get_readfd()
API:gst_rtsp_connection_get_writefd()
 2009-03-06 18:54:57 +01:00
Julien Moutte
d45b27d92d Fix build on Mac OS X 2009-03-06 10:37:38 +01:00
Wim Taymans
f69a3d953a rtsp: fix parsing of 'now-' ranges. 
--
 2009-03-05 13:48:37 +01:00
Wim Taymans
bcaec3d907 rtsp: do some more cleanup in _close 
Do som more cleanup in gst_rtsp_connection_close() so that it's back into the
unconnected state as it was allocated.
 2009-03-04 16:24:01 +01:00
Wim Taymans
629f2dcee4 rtsp: fix the memory management of the url 
Constify the url parameter in _create.
Make a copy of the url stored in the connection.
Free the url when the connection is freed.
 2009-03-04 16:11:20 +01:00
Wim Taymans
b6d7a1dc03 RTSP: Add support for server tunneling 
Save the tunnelid in the connection. Add a method to retrieve the tunnelid so
that a server can store and match the id against other tunnel requests.

Fix the URI in the tunnel requests so that they contain the absolute uri and the
query string if any instead of just the hostname.

Transparently base64 decode the input stream when tunneling.

Add method to set the connection ip address so that it can be included in the
tunnel response.

Add method to connect the two tunnel requests.

Add two callbacks for the async mode to notify a tunnel start and tunnel
complete event.

Add method to reset the watch after the connection has been tunneled.

Various little refactoring to make more stuff reusable.

API: RTSP::gst_rtsp_connection_set_ip()
API: RTSP::gst_rtsp_connection_get_tunnelid()
API: RTSP::gst_rtsp_connection_do_tunnel()
API: RTSP::gst_rtsp_watch_reset()
 2009-03-04 12:21:29 +01:00
Wim Taymans
3b6e9fc870 rtsp: add new defines for tunneling 
Add two more result codes for tunneling support.
 2009-03-04 12:18:00 +01:00
Wim Taymans
9ea1240910 rtsp: remove , from last enum member 
Remove , from last enum member to improve compatibility with other compilers.
 2009-03-04 12:12:06 +01:00
Wim Taymans
9045d210b2 rtsp: remove debugging g_message 
--
 2009-03-02 16:13:33 +01:00
Wim Taymans
fbc4f2d4fe RTSP: add support for Quicktime tunneled RTSP 
Add support for tunneling RTSP over HTTP.
Fix documentation some more.
See also #573173.

API: RTSP:gst_rtsp_connection_is_tunneled()
API: RTSP:gst_rtsp_connection_set_tunneled()
 2009-03-02 16:03:49 +01:00
Wim Taymans
40db590e71 RTSP: parse rtsph uris as RTSP tunneled over HTTP 
Add transport define for RTSP tunneled over HTTP.

Parse rtsph:// uris as tunneled HTTP over TCP.

API: GstRTSPLowerTrans::GST_RTSP_LOWER_TRANS_HTTP

See also #573173.
 2009-03-02 15:48:56 +01:00
Wim Taymans
4664fe40bc rtsp: add _get_url method and separate sockets 
Add gst_rtsp_connection_get_url() method.

Reserve space for 2 sockets, one for reading and one for writing. Use socket
pointers to select the read and write sockets. This should allow us to implement
tunneling over HTTP soon.

API: RTSP::gst_rtsp_connection_get_url()
 2009-03-02 10:58:49 +01:00
Tim-Philipp Müller
0a835bc9a3 app: force automatic rebuild of gstapp-marshal.[ch] after previous change 
The previous change to appsrc/appsink requires people to 'make clean'
to get the marshallers rebuilt (causing a build failure otherwise).
Change some lines in the .list file around to force a rebuild of
these files automatically.
 2009-03-01 18:31:17 +00:00
LRN
e5d2d32bba rtspconnection: Use correct types for some functions on Win32 
Fixes bug #573529.
 2009-02-28 19:35:33 +01:00
Edward Hervey
ed013753c0 rtspconnection: Fix warning about using unitialized value. 2009-02-28 13:11:59 +01:00
Edward Hervey
6f73427aa6 riff: Add more codec mappings. 
This comes mostly from a review of ffmpeg/libavformat/riff.c
 2009-02-28 12:41:28 +01:00
Stefan Kost
4e4f922d7a rtsprange: don't leak the range in case of parsing error. 
Free the gstRTSPTimeRange if we don't return it. Also simplify
gst_rtsp_range_free() as it is valid to pass NULL to g_free().
 2009-02-26 18:01:05 +02:00
Wim Taymans
c4036dd701 app: add callbacks to appsrc, cleanups 
Add a uri handler to appsink.
don't emit signals when we have installed callbacks on appsink.

Add callbacks to appsrc to replace the signals.
Add property to disable callbacks in appsrc, default to TRUE for backwards
compatibility but disable when callbacks are installed.

API: GstAppSrc::emit-signals
API: GstAppSrc::gst_app_src_set_emit_signals()
API: GstAppSrc::gst_app_src_get_emit_signals()
API: GstAppSrc::gst_app_src_set_callbacks()
 2009-02-26 16:44:53 +01:00
Wim Taymans
661f2da6e0 Appsink: add padding for callbacks + docs 
Add some padding to the callbacks structure just to be safe.

Remove the now invisible marshaller methods from the docs.

Fix a comment in the unit test.
 2009-02-26 11:42:44 +01:00
Stefan Kost
58695d78f9 docs: fix newly added interlace constants and plug holes in video format docs 2009-02-26 10:09:59 +02:00
Stefan Kost
251e4d160a docs: don't put random stuff in tags. 
Tags like Since: or Returns: can only be followed by more tags. gtk-doc has no
tag to append text again to the documentation body.
 2009-02-26 10:09:59 +02:00
Tim-Philipp Müller
07d2dbfdfe app: add win32 .def file and only export functions we want exported 
Add a .def file for win32 builds (and make check-exports).
Fix LDFLAGS in Makefile.am, so the usual export regexps are used (fixes #573165).
Make sure private marshaller functions aren't exported by prefixing them with __gst;
also rename gst_app_marshal_OBJECT__VOID to _BUFFER__VOID, make it static and add
a comment why we're not using glib-genmarshal for this one.
 2009-02-25 19:50:00 +00:00
Peter Kjellerstedt
2fe8e4c1de Fixed a typo. 2009-02-25 16:25:33 +01:00
Peter Kjellerstedt
a038a8d46d rtsp, multifdsink: Unify the use of union gst_sockaddr. 2009-02-25 15:45:50 +01:00
Tim-Philipp Müller
3d88a5b985 riff: add fourcc for mpeg2-in-avi (as produced by mencoder) 
Fixes: #565777
 2009-02-25 11:13:01 +00:00
Edward Hervey
e57073b6f9 Riff: Add fourcc for mpeg1-in-avi (as produced by mencoder) 2009-02-25 08:05:58 +01:00
Garret D'Amore
b8af1223db mixer interface: Add flags to enhance mixer interfaces 
This patch adds a few flags to the mixer and mixerctrl interface to
better support OSSv4 (and potentially other backends).

Patch By: Garret D'Amore <garrett.damore@sun.com>
Signed-Off-By: Jan Schmidt <jan.schmidt@sun.com>

API: GST_MIXER_FLAG_HAS_WHITELIST, GST_MIXER_FLAG_GROUPING,
API: GST_MIXER_TRACK_NO_RECORD, GST_MIXER_TRACK_NO_MUTE,
API: GST_MIXER_TRACK_WHITELIST
 2009-02-24 17:23:58 +00:00
Jan Schmidt
94791df88d rtsp: Fix a strict aliasing warning 
Fix strict aliasing warnings from casting a sockaddr_storage and
using it as a sockaddr_in6. Use a union instead.
 2009-02-24 16:49:40 +00:00
Wim Taymans
bb5e2d3f56 Match WSAStartup and WSACleanup correctly 
Don't randomly call WSAStartup and WSACleanup but instead call the startup when
we create a connection and cleanup when we free it again. Because the internal
datastructure is refcounted, this should not cause any refcounting leaks when
the connection is managed correctly.
Fixes #562794.
 2009-02-24 12:11:00 +01:00
Wim Taymans
6e560ae5d8 Add method for handling server requests 
Add a receive_request so that extensions can react to server requests.
 2009-02-23 10:57:08 +01:00
Sebastian Dröge
d659e8353d tagdemux: Unref the actual buffer instead of the memory address of the buffer 2009-02-22 19:12:00 +01:00
Edward Hervey
5ce5433152 libs/video: Fix gst_video_format_new_caps* functions. 
Only add a 'interlaced=True' property to caps *IF* it is interlaced, else
don't add anything.
 2009-02-22 13:42:33 +01:00
Wim Taymans
15cd839f81 Improve key/value parsing 
Improve header field parsing by keeping a ref to the key/value instead of
copying it into a local variable.
 2009-02-20 17:26:40 +01:00
Wim Taymans
bb4310203a Add trailing \0 to message length 
We always put a trailing 0 at the end of the message body. Reflect this fact in
the length of the message.
 2009-02-20 12:35:53 +01:00
Wim Taymans
0ffd5e703a Don't parse headers for data messages 
Don't try to parse the headers on a data message because they don't have
headers.
 2009-02-20 09:52:16 +01:00
Edward Hervey
a490b3f2dd video: Fix 'Since' tags 2009-02-19 17:40:45 +01:00
Edward Hervey
c44b067817 video: Add flags for interlaced video along with convenience methods for interlaced caps. 
These three flags allow all know combinations of interlaced formats. They should
only be used when the caps contain 'interlaced=True'.

Fixes #163577 (yes, it's a 4 year old bug).
 2009-02-19 16:11:44 +01:00
Wim Taymans
f187ffddce Make RTSPConnection opaque and rename RTSPChannel 
Make the RTSPConnection object opaque so that we can extend it in the future.

Rename GstRTSPChannel to GstRTSPWatch to avoid confusing with the RTSP channels.
 2009-02-19 15:55:07 +01:00
Edward Hervey
02f9079d6b Add some more mappings for h264 in riff 2009-02-19 13:24:39 +01:00
Wim Taymans
e5d8551552 Add method to install callbacks on appsink 
Based on pacth by Martin Samuelsson <martin dot samuelsson at axis dot com>
Fixes #571299.

Add gst_app_sink_set_callbacks() to install a set of callbacks. This is a more
performant alternative to connecting to the signals.

Add a unit test for appsink.

Clean up some of the appsink docs.

API: GstAppSink::gst_app_sink_set_callbacks()
 2009-02-19 10:44:31 +01:00
Wim Taymans
a2f04c8f61 Add RTSP accept method 
Add a method to accept a connection on a socket and create a GstRTSPConnection
for it.

API: gst_rtsp_connection_accept()
 2009-02-18 18:46:35 +01:00
Wim Taymans
a6d75bd33c Add RTSP channel object for async io 
Add a GstRTSPChannel object that wraps a GSource around the RTSP connection so
that the connection can be monitored from a maincontext. This allows us to
operate in ASYNC mode, which is handy when building a server.

Rework the old code to use the async code under the hood.

API: gst_rtsp_channel_new()
API: gst_rtsp_channel_unref()
API: gst_rtsp_channel_attach()
API: gst_rtsp_channel_queue_message()
 2009-02-18 17:42:59 +01:00
Tim-Philipp Müller
a624df17c4 tagdemux: don't abort when downstream pulls a buffer of size 0 
Pulling a 0-sized buffer is allowed, and we should handle this correctly instead of
aborting. Fixes #571009 (wma file with ID3v2 tag).
 2009-02-12 09:18:20 +00:00
Tim-Philipp Müller
1fedfec220 riff: error out on nonsensical chunk sizes instead of aborting 
When encountering a nonsensical chunk size such as (guint)-1, error out cleanly instead of
continuing and trying to g_memdup() 4GB of data that doesn't exist, which will either abort
in g_malloc() or crash.

Fixes #553295, crash with fuzzed AVI file.
 2009-02-11 16:58:18 +00:00
Peter Kjellerstedt
430eea3016 gstrtspmessage: Minor documentation correction. 
Corrected documentation about what needs to be freed after calling
gst_rtsp_message_new(), gst_rtsp_message_new_request(),
gst_rtsp_message_new_response() and gst_rtsp_message_new_data().
 2009-02-10 17:37:06 +01:00
Wim Taymans
76112f9f04 RTSPRange: Add method to serialize ranges 
Add gst_rtsp_range_to_string() to serialize a GstRTSPRange to a string that can
be used by a server.
API: GstRTSPRange::gst_rtsp_range_to_string()
 2009-02-04 17:03:52 +01:00
Wim Taymans
4bb5722f1a GstRTSPUrl: Add some const to methods 
Add const to the methods that do not modify the object.
 2009-02-04 13:16:48 +01:00
Wim Taymans
ad1dea3122 Add more g_return_if_fail() calls 
Check that we have a valid file descriptor before entering certain functions in
order to avoid undesirable situations.
Add some more debugging in the connect method.
 2009-02-04 11:18:31 +01:00
Tim-Philipp Müller
95d6fb0501 pbutils: remove duplicate detail strings when calling the external codec installer 
It doesn't make sense to ask installers for the same codec or element twice, so filter out duplicate requests before calling the external helper script and make the unit test check this works right. Fixes #567636.
 2009-02-02 17:34:23 +00:00
Stefan Kost
486fe43cb9 Add a FIXME 0.11. Make the log message a bit more detailed and add comments. 2009-02-02 18:05:42 +02:00
Wim Taymans
35cec4c006 Fix string leak in rtspmessage 
when we remove a header field from a message we must free the value associated
with the key to avoid a memory leak.
 2009-02-02 10:09:07 +01:00
Stefan Kost
950d0c0a7d Link to the class, as we can't link to the members yet. 2009-01-31 18:44:32 +02:00
Wim Taymans
6f3511bfb6 fix some typos 
Fix some typos in the doc string of the new
gst_rtsp_options_as_string() method.
 2009-01-29 14:00:30 +01:00
Wim Taymans
484a025f6d Add new RTSP message method to set header 
Add gst_rtsp_message_take_header() that takes ownership of the passed header
value. This allows us to avoid an allocations and memory copy in some
situations.
API: GstRTSPMessage::gst_rtsp_message_take_header()
 2009-01-29 11:55:10 +01:00
Wim Taymans
e8bd8cab41 Add method to serialize RTSP options 
Add gst_rtsp_options_as_text() method to serialize a set of RTSP options to a
string.
API: GstRTSP::gst_rtsp_options_as_text()
 2009-01-28 11:48:01 +01:00
Jan Schmidt
63c9ede3d0 Extend and clean up git ignores 2009-01-23 23:16:11 +00:00
Wim Taymans
a7f2540f77 Add more codec ids for RIFF formats 
Handle codec ID for various other AAC formats.
Sync the list of possible codec ids with that of ffmpeg.
Fixes #567255
 2009-01-23 11:33:29 +01:00
Wim Taymans
26256b95c8 Reset queued_bytes counter when flushing 
Set the amount of queued bytes in the internal queue back to 0 when we clear the
queue.
Fixes #567982
 2009-01-23 11:11:31 +01:00
Wim Taymans
509f561ef3 Merge branch 'master' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-base 2009-01-22 13:12:02 +01:00
Sebastian Dröge
2e8f9921c9 Reduce the number of allocations for creating FFT contexts 
Reduce the number of allocations from 2 to 1 for every FFT
context by allocating enough memory for the FFT context
and passing parts of it to the kissfft allocation functions.
 2009-01-22 12:54:35 +01:00
Wim Taymans
9ce042e2a7 Avoid overflows in the padding checks by doing the check slightly 
differently.
Add a unit test to check for correct behaviour.
 2009-01-21 13:09:29 +01:00
Sebastian Dröge
4d3ff205be gst-libs/gst/fft/: Use correct struct alignment everywhere to prevent unaligned memory accesses, resulting in SIGBUS ... 
Original commit message from CVS:
* gst-libs/gst/fft/_kiss_fft_guts_f32.h:
* gst-libs/gst/fft/_kiss_fft_guts_f64.h:
* gst-libs/gst/fft/_kiss_fft_guts_s16.h:
* gst-libs/gst/fft/_kiss_fft_guts_s32.h:
* gst-libs/gst/fft/kiss_fftr_f32.c: (kiss_fftr_f32_alloc):
* gst-libs/gst/fft/kiss_fftr_f64.c: (kiss_fftr_f64_alloc):
* gst-libs/gst/fft/kiss_fftr_s16.c: (kiss_fftr_s16_alloc):
* gst-libs/gst/fft/kiss_fftr_s32.c: (kiss_fftr_s32_alloc):
Use correct struct alignment everywhere to prevent unaligned
memory accesses, resulting in SIGBUS on sparc and probably others.
Fixes bug #500833.
 2009-01-16 11:44:04 +00:00
Sebastian Dröge
98ea758763 gst-libs/gst/tag/gsttagdemux.c: Forward unknown events upstream to allow latency configuration. 
Original commit message from CVS:
* gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_srcpad_event):
Forward unknown events upstream to allow latency configuration.
Fixes bug #567960.
 2009-01-16 11:40:02 +00:00
Jan Schmidt
80ac3b565e gst-libs/gst/app/gstappsink.c: Store the returned signal id in the right slot when registering the pull-buffer signal. 
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c:
Store the returned signal id in the right slot when
registering the pull-buffer signal.
Fixes #567168
Spotted by: Thomas Vander Stichele  <thomas at apestaart dot org>
 2009-01-09 23:13:17 +00:00
Tim-Philipp Müller
d629c9fc17 gst-libs/gst/interfaces/mixer.c: Small docs addition to clarify that one really mustn't free the constant GList retur... 
Original commit message from CVS:
* gst-libs/gst/interfaces/mixer.c:
Small docs addition to clarify that one really mustn't free
the constant GList returned (#566812).
 2009-01-09 17:17:50 +00:00
Wim Taymans
1f6297f051 Add GType for GstRTSPUrl and expose a copy function because we can. 
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtsp/gstrtspurl.c: (register_rtsp_url_type),
(gst_rtsp_url_get_type), (gst_rtsp_url_copy):
* gst-libs/gst/rtsp/gstrtspurl.h:
* win32/common/libgstrtsp.def:
Add GType for GstRTSPUrl and expose a copy function because we can.
API: gst_rtsp_url_copy()
Fixes #567027.
 2009-01-08 17:18:24 +00:00
Sebastian Dröge
ba03cb6080 gst-libs/gst/cdda/gstcddabasesrc.*: Make the GType of GstCDDABaseSrcMode public for bindings. 
Original commit message from CVS:
* gst-libs/gst/cdda/gstcddabasesrc.c:
* gst-libs/gst/cdda/gstcddabasesrc.h:
Make the GType of GstCDDABaseSrcMode public for bindings.
Fixes bug #566837.
 2009-01-07 10:50:15 +00:00
José Alburquerque
7431789249 gst-libs/gst/audio/gstaudioclock.*: Make gst_audio_clock_new use const gchar* to ease the wrapping of 
Original commit message from CVS:
Patch by: José Alburquerque <jaalburqu svn gnome org>
* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_new):
* gst-libs/gst/audio/gstaudioclock.h:
Make gst_audio_clock_new use const gchar* to ease the wrapping of
C++ bindings. Fixes #566723.
 2009-01-06 17:30:31 +00:00
Tim-Philipp Müller
ada70bb159 gst-libs/gst/app/: Make debug categories static. Use _element_class_set_details_simple(). 
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c:
* gst-libs/gst/app/gstappsink.h:
* gst-libs/gst/app/gstappsrc.c:
* gst-libs/gst/app/gstappsrc.h:
Make debug categories static. Use _element_class_set_details_simple().
 2009-01-06 11:10:29 +00:00
Tim-Philipp Müller
d2b82026c8 gst-libs/gst/app/: Move private data into a private instance struct. Add padding to instance and class structures exp... 
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (_GstAppSinkPrivate),
(gst_app_sink_class_init), (gst_app_sink_init),
(gst_app_sink_dispose), (gst_app_sink_finalize),
(gst_app_sink_unlock_start), (gst_app_sink_unlock_stop),
(gst_app_sink_flush_unlocked), (gst_app_sink_start),
(gst_app_sink_stop), (gst_app_sink_event), (gst_app_sink_preroll),
(gst_app_sink_render), (gst_app_sink_getcaps),
(gst_app_sink_set_caps), (gst_app_sink_get_caps),
(gst_app_sink_is_eos), (gst_app_sink_set_emit_signals),
(gst_app_sink_get_emit_signals), (gst_app_sink_set_max_buffers),
(gst_app_sink_get_max_buffers), (gst_app_sink_set_drop),
(gst_app_sink_get_drop), (gst_app_sink_pull_preroll),
(gst_app_sink_pull_buffer)::
* gst-libs/gst/app/gstappsink.h: (GstAppSinkPrivate), (_GstAppSink)::
* gst-libs/gst/app/gstappsrc.c: (_GstAppSrcPrivate),
(gst_app_src_class_init), (gst_app_src_init),
(gst_app_src_flush_queued), (gst_app_src_dispose),
(gst_app_src_finalize), (gst_app_src_set_property),
(gst_app_src_get_property), (gst_app_src_unlock),
(gst_app_src_unlock_stop), (gst_app_src_start), (gst_app_src_stop),
(gst_app_src_is_seekable), (gst_app_src_check_get_range),
(gst_app_src_query), (gst_app_src_do_seek), (gst_app_src_create),
(gst_app_src_set_caps), (gst_app_src_get_caps),
(gst_app_src_set_size), (gst_app_src_get_size),
(gst_app_src_set_stream_type), (gst_app_src_get_stream_type),
(gst_app_src_set_max_bytes), (gst_app_src_get_max_bytes),
(gst_app_src_set_latencies), (gst_app_src_set_latency),
(gst_app_src_get_latency), (gst_app_src_push_buffer_full),
(gst_app_src_push_buffer_action), (gst_app_src_end_of_stream)::
* gst-libs/gst/app/gstappsrc.h: (GstAppSrcPrivate)::
Move private data into a private instance struct. Add padding to
instance and class structures exposed in public headers. Add
Since markers to the gtk-doc blurbs (#566750).
 2009-01-06 10:56:45 +00:00
Jan Schmidt
1b2dc5f3a8 gst-libs/gst/video/: Fix up build flags and include statement for the new generated enumtypes files, to fix dist. 
Original commit message from CVS:
* gst-libs/gst/video/Makefile.am:
* gst-libs/gst/video/video.h:
Fix up build flags and include statement for the new generated
enumtypes files, to fix dist.
 2009-01-06 10:16:16 +00:00
Jan Schmidt
08393941a8 Move AppSrc/AppSink from gst-plugins-bad. Fixes #564421 
Original commit message from CVS:
* configure.ac:
* docs/libs/Makefile.am:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.interfaces:
* docs/plugins/gst-plugins-base-plugins.prerequisites:
* docs/plugins/gst-plugins-base-plugins.signals:
* docs/plugins/inspect/plugin-app.xml:
* gst-libs/gst/Makefile.am:
* gst-libs/gst/app/gstappsink.c:
* gst-libs/gst/app/gstappsrc.c:
* tests/examples/Makefile.am:
* tests/examples/app/Makefile.am:
Move AppSrc/AppSink from gst-plugins-bad. Fixes #564421
 2009-01-05 23:04:57 +00:00
Wim Taymans
0a4c1bc64c gst-libs/gst/audio/gstbaseaudiosink.c: Avoid holding the OBJECT_LOCK when calling ringbuffer functions that take the ... 
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_change_state):
Avoid holding the OBJECT_LOCK when calling ringbuffer functions that
take the ringbuffer lock because rinbuffer lock > OBJECT_LOCK. We can do
this because the async_play method is deprecated and usually not called
anymore.
 2009-01-05 17:13:13 +00:00