mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-25 09:40:37 +00:00
Code Issues 1 Projects Releases Wiki Activity
7178 commits 44 branches 1510 tags 330 MiB
Commit graph

1556 commits

Author SHA1 Message Date
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