Commit graph

1492 commits

Author SHA1 Message Date
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