Commit graph

7897 commits

Author SHA1 Message Date
Sebastian Dröge
b357cb9d2a fft: Add gobject-introspection support 2009-09-05 11:09:45 +02:00
Sebastian Dröge
a5f7c699ca cdda: Add gobject-introspection support
This is disabled for now until gobject-introspection is fixed
2009-09-05 11:09:39 +02:00
Sebastian Dröge
403f353bba audio: Add gobject-introspection support 2009-09-05 11:09:33 +02:00
Sebastian Dröge
61ae0059a4 app: Add gobject-introspection support 2009-09-05 11:09:28 +02:00
Wim Taymans
7a7663476f audiortppay: add some debugging 2009-09-03 18:53:19 +02:00
Wim Taymans
c1db9ebb20 audiortppay: handle gaps
Add various conversion functions between time<->bytes<->rtptime that will be
used later on.
Refactor the min/max packet length code so that it can be used for both
sample/frame based payloaders. Cache the returned values.
code cleanups.
When we discover a DISCONT buffer, make the outgoing RTP timestamps have the
same gap as the GStreamer timestamps gap.
2009-09-03 17:59:00 +02:00
Wim Taymans
3a3c6f309c audiortppay: fix frame duration calculations
Fix the calculation of the frame duration and rtp timestamps.
Add some debugging
2009-09-03 17:59:00 +02:00
Wim Taymans
bfc19462bb rtppay: add some debugging 2009-09-03 17:59:00 +02:00
Wim Taymans
bb91a7b47c audiortppay: use offsets for RTP timestamps
Have a custom sample/frame function to generate an offset that the base class
will use for generating RTP timestamps. This results in perfect RTP timestamps
on the output buffers.
Refactor setting metadata on output buffers.
Add some more functionality to _flush().
Handle DISCONT on the input buffers and set the marker bit and DISCONT flag on
the next outgoing buffer.
Flush the pending data on EOS.
2009-09-03 17:58:59 +02:00
Wim Taymans
c1ae0a2003 audiortppay: move function around 2009-09-03 17:58:59 +02:00
Wim Taymans
5808041f44 audiortppay: fix sample duration calculation 2009-09-03 17:58:59 +02:00
Wim Taymans
299ab7be0e audiortppay: more refactoring
Unify the sample/frame buffer handling code by making the functions plugable.
2009-09-03 17:58:59 +02:00
Wim Taymans
fb5037f727 audiortppayload: refactor some more
Refactor getting the packet min/max size and alignment code.
Refactor converting bytes to time.
change some variable to something shorter.
2009-09-03 17:58:59 +02:00
Wim Taymans
1c6b71af03 audiortppayload: refactor and cleanup
Always use the adapter when we need to fragment the incomming buffer. Use more
modern adapter functions to avoid malloc and memcpy. The overall result is that
the code looks cleaner while it should be equally fast and in some case avoid a
memcpy and malloc.
Use the adapter timestamping functions for more precise timestamps in case of
weird disconts.
Cache some values instead of recalculating them.
Add gst_base_rtp_audio_payload_flush() to flush a certain amount of bytes from
the internal adapter.

API: GstBaseRTPAudioPayload::gst_base_rtp_audio_payload_flush()
2009-09-03 17:58:59 +02:00
Wim Taymans
50b9640d01 basertppay: add property to disable perfect RTP time
Add a property to disable the generation of perfect RTP timestamps. By default
it is active.

API: GstBaseRTPPayload::perfect-rtptime
2009-09-03 11:29:23 +02:00
Wim Taymans
3a4edea56d basertppay: allow subclasses to influence RTP time
Allow subclasses to use the OFFSET field on RTP buffers to influence the way in
which RTP timestamps are generated. Usually timestamps are created from the
GStreamer timestamps on the buffer, which could result in imperfect RTP
timestamps.
2009-09-03 11:15:20 +02:00
Wim Taymans
5a479669d4 basertppay: add macro to cast 2009-09-03 11:15:20 +02:00
Wim Taymans
bc3c8a1564 audiopayload: code cleanups 2009-09-03 11:15:20 +02:00
Wim Taymans
3c29efa692 audiortppayload: don't check adapter
the adapter is never NULL so we don't need to check it.
Use _scale functions to avoid overflows.
2009-09-03 11:15:20 +02:00
Jonas Holmberg
ec91d508af basertppayload: Make instance init faster by not reading /dev/urandom 3 times
... which is the default seed when creating a new GRand. Because
GLib in older versions used buffered IO this would take a lot of time.

Instead use the global GRand for getting random numbers and keep the
three instance GRand for backward compatibility with a simple seed.

Fixes bug #593284.
2009-09-01 10:39:52 +02:00
Wim Taymans
008c760b6b cddabasesrc: safely handle the indexes 2009-08-28 19:06:57 +02:00
Wim Taymans
e40b262ab7 basertppayload: whitespace fixes. 2009-08-28 14:09:02 +02:00
Sebastian Dröge
72f3587f04 riff: Add support for AVF files
AVF is valid RIFF but has AVF0 has first fourcc instead of RIFF.

Fixes bug #593117.
2009-08-26 09:10:19 +02:00
Peter Kjellerstedt
8ce3612b71 rtsp: Mark Transport as supporting multiple values. 2009-08-24 14:39:16 +02:00
Peter Kjellerstedt
2882c22d95 rtsp: Added missing Since tags. 2009-08-24 13:58:50 +02:00
Eero Nurkkala
8ad8591e41 ringbuffer: Improve audiosink startup performance
When we start the ringbuffer, immediatly continue processing samples if the
writer prepared some for us.

Fixes #545807
2009-08-24 13:30:11 +02:00
Peter Kjellerstedt
066f9be5c9 rtsp: Added new API for sending using GstRTSPWatch.
The new API to send messages using GstRTSPWatch will first try to send the
message immediately. Then, if that failed (or the message was not sent
fully), it will queue the remaining message for later delivery. This avoids
unnecessary context switches, and makes it possible to keep track of
whether the connection is blocked (the unblocking of the connection is
indicated by the reception of the message_sent signal).

This also deprecates the old API (gst_rtsp_watch_queue_data() and
gst_rtsp_watch_queue_message().)

API: gst_rtsp_watch_write_data()
API: gst_rtsp_watch_send_message()
2009-08-24 13:19:46 +02:00
Peter Kjellerstedt
0af04aa4a8 rtsp: Made gst_rtsp_watch_queue_data() thread safe. 2009-08-24 13:19:46 +02:00
Peter Kjellerstedt
fb3b761af5 rtsp: Added gst_rtsp_connection_set_http_mode().
With gst_rtsp_connection_set_http_mode() it is possible to tell the
connection whether to allow HTTP messages to be supported. By enabling HTTP
support the automatic HTTP tunnel support will also be disabled.

API: gst_rtsp_connection_set_http_mode()
2009-08-24 13:19:46 +02:00
Peter Kjellerstedt
d5b4b5d8af rtsp: Allow gst_rtsp_connection_do_tunnel() to just setup decoding context.
If the second connection passed to gst_rtsp_connection_do_tunnel() is NULL
then just setup the base64 decoding context for the first connection.
2009-08-24 13:19:46 +02:00
Peter Kjellerstedt
01d98fdb5d rtsp: Write as much as possible in gst_rtsp_source_dispatch().
Try to write as much as possible if there are multiple messages queued.
2009-08-24 13:19:45 +02:00
Peter Kjellerstedt
e5ec74c7a9 rtsp: Add error_full callback to GstRTSPWatchFuncs.
The error_full callback is similar to the error callback, but allows for
better error handling. For read errors a partial message is provided to
help an RTSP server generate a more correct error response, and for write
errors the write queue id of the failed message is returned.
2009-08-24 13:19:45 +02:00
Peter Kjellerstedt
ab8bea4555 rtsp: Made read_line() support LWS.
Rewrote read_line() to support LWS (Line White Space), the method used by
RTSP (and HTTP) to break long lines. Also added support for \r and \n as
line endings (in addition to the official \r\n).
2009-08-24 13:19:45 +02:00
Peter Kjellerstedt
607209f121 rtsp: Do not split headers which should not be split.
From RFC 2068 section 4.2: "Multiple message-header fields with the same
field-name may be present in a message if and only if the entire
field-value for that header field is defined as a comma-separated list
[i.e., #(values)]." This means that we should not split other headers which
may contain a comma, e.g., Range and Date.
2009-08-24 13:19:45 +02:00
Peter Kjellerstedt
08d3fe8561 rtsp: Parse WWW-Authenticate headers correctly.
Due to the odd syntax for WWW-Authenticate (and Proxy-Authenticate) which
allows commas both to separate between multiple challenges, and within the
challenges themself, we need to take some extra care to split these headers
correctly.
2009-08-24 13:19:45 +02:00
Peter Kjellerstedt
efc8901a39 rtsp: Improve parse_line().
Make parse_line() handle keys with multiple values on one line correctly.
2009-08-24 13:19:45 +02:00
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
Edward Hervey
70a35897fb gst-libs/gst/tag/gsttagdemux.*: Add GType for GstTagDemuxResult enum.
Original commit message from CVS:
* gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_result_get_type):
* gst-libs/gst/tag/gsttagdemux.h:
Add GType for GstTagDemuxResult enum.
2008-12-31 13:31:55 +00:00
Edward Hervey
98ad43fcdd gst-libs/gst/video/: Add glib-mkenum for GstVideoFormat enum GTYPE auto-generation.
Original commit message from CVS:
* gst-libs/gst/video/Makefile.am:
* gst-libs/gst/video/video.h:
Add glib-mkenum for GstVideoFormat enum GTYPE auto-generation.
This will help bindings to use it.
2008-12-31 13:01:30 +00:00
Edward Hervey
e2fcc71650 Switch glib-mkenum for gst-libs/gst/audio from multichannel- to audio- in order to wrap all enums declarations of tha...
Original commit message from CVS:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/audio/audio.c:
* gst-libs/gst/audio/multichannel.h:
* gst-libs/gst/audio/testchannels.c:
* win32/MANIFEST:
* win32/common/audio-enumtypes.c:
(gst_audio_channel_position_get_type),
(gst_ring_buffer_state_get_type),
(gst_ring_buffer_seg_state_get_type),
(gst_buffer_format_type_get_type), (gst_buffer_format_get_type):
* win32/common/audio-enumtypes.h:
* win32/common/multichannel-enumtypes.c:
* win32/common/multichannel-enumtypes.h:
* win32/vs6/grammar.dsp:
* win32/vs6/libgstaudio.dsp:
* win32/vs7/libgstaudio.vcproj:
* win32/vs8/libgstaudio.vcproj:
Switch glib-mkenum for gst-libs/gst/audio from multichannel- to
audio- in order to wrap all enums declarations of that library.
This modification should not matter since that header file is not a
public header (it will be included by public headers).
Modify win32 crap^Wfiles accordingly.
2008-12-31 11:20:26 +00:00
Edward Hervey
20adaa1328 gst-libs/gst/audio/: Complete Sebastien's commit from the 13th by exporting the _slave_method_get_type() methods.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.h:
* gst-libs/gst/audio/gstbaseaudiosink.h:
Complete Sebastien's commit from the 13th by exporting the
_slave_method_get_type() methods.
2008-12-30 17:55:07 +00:00
Wim Taymans
0ab6c0fbc0 gst-libs/gst/app/gstappsrc.*: Add properties and methods to configure and retrieve the min and max latencies.
Original commit message from CVS:
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
(gst_app_src_init), (gst_app_src_set_property),
(gst_app_src_get_property), (gst_app_src_query),
(gst_app_src_set_latencies), (gst_app_src_set_latency),
(gst_app_src_get_latency), (gst_app_src_push_buffer_full):
* gst-libs/gst/app/gstappsrc.h:
Add properties and methods to configure and retrieve the min and max
latencies.
2008-12-29 16:45:20 +00:00
Wim Taymans
a579eba73d gst-libs/gst/audio/gstbaseaudiosink.c: Pause the write thread before deactivating and releasing the ringbuffer to avo...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_setcaps):
Pause the write thread before deactivating and releasing the ringbuffer
to avoid a deadlock when we do gapless playback with different sample
rates in playbin2.  Fixes #564929.
2008-12-20 12:45:03 +00:00
Sebastian Dröge
4ed1f5d6fd gst-libs/gst/audio/gstbaseaudiosrc.c: Make GstAudioSrcSlaveMethod get_type() function non-static as it's public now.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
Make GstAudioSrcSlaveMethod get_type() function non-static
as it's public now.
* win32/common/libgstaudio.def:
* win32/common/libgstnetbuffer.def:
Add some missing functions to the list of exported symbols.
2008-12-19 13:03:00 +00:00
Andrew Feren
a628077e96 gst-libs/gst/netbuffer/gstnetbuffer.*: Make gst_netaddress_get_ip4_address fail for v6 addresses.
Original commit message from CVS:
Patch by: Andrew Feren <acferen at yahoo dot com>
* gst-libs/gst/netbuffer/gstnetbuffer.c:
(gst_netaddress_get_ip4_address), (gst_netaddress_get_ip6_address),
(gst_netaddress_get_address_bytes),
(gst_netaddress_set_address_bytes):
* gst-libs/gst/netbuffer/gstnetbuffer.h:
Make gst_netaddress_get_ip4_address fail for v6 addresses.
Make gst_netaddress_get_ip6_address either fail or return the v4
address as a transitional v6 address.
Add two convenience functions:
API: gst_netaddress_get_address_bytes()
API: gst_netaddress_set_address_bytes()
Fixes #564896.
2008-12-18 12:37:33 +00:00
Wim Taymans
8567ee2149 Add appsrc and appsink documentation.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* gst-libs/gst/app/gstappsink.c:
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init):
Add appsrc and appsink documentation.
2008-12-17 13:51:46 +00:00
Wim Taymans
24685b5df0 examples/app/: Fix example to unref after emiting the push-buffer action.
Original commit message from CVS:
* examples/app/appsrc-ra.c: (feed_data):
* examples/app/appsrc-seekable.c: (feed_data):
* examples/app/appsrc-stream.c: (read_data):
* examples/app/appsrc-stream2.c: (feed_data):
Fix example to unref after emiting the push-buffer action.
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
(gst_app_src_push_buffer_full), (gst_app_src_push_buffer),
(gst_app_src_push_buffer_action):
Don't take the ref on the buffer in push-buffer action because it's too
awkward for bindings. Fixes #564482.
2008-12-15 12:02:26 +00:00
Sebastian Dröge
04d9ff9a24 gst-libs/gst/audio/: API: Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the public API. This is needed for the C...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_slave_method_get_type),
(gst_base_audio_sink_class_init):
* gst-libs/gst/audio/gstbaseaudiosink.h:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_slave_method_get_type),
(gst_base_audio_src_class_init):
* gst-libs/gst/audio/gstbaseaudiosrc.h:
API: Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the
public API. This is needed for the C++ bindings to be able
to use this base classes. Fixes bug #564200, #564206.
2008-12-13 06:57:09 +00:00
Edward Hervey
c5ae184910 gst-libs/gst/cdda/gstcddabasesrc.c: Remove erroneous gst_buffer_ref().
Original commit message from CVS:
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_handle_event):
Remove erroneous gst_buffer_ref().
* tests/check/libs/rtp.c: (GST_START_TEST):
Don't forget to unref the buffer once you're done with it.
2008-12-12 19:41:28 +00:00
Edward Hervey
c4295a07b9 gst-libs/gst/riff/riff-media.c: Add mapping for VP6 in avi/riff.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
(gst_riff_create_video_template_caps):
Add mapping for VP6 in avi/riff.
2008-12-12 07:15:22 +00:00
Luis Menina
a4493595a6 gst/: Include glib.h instead of a specific GLib header. Including single
Original commit message from CVS:
Patch by: Luis Menina <liberforce at freeside dot fr>
* gst-libs/gst/floatcast/floatcast.h:
* gst/typefind/gsttypefindfunctions.c:
Include glib.h instead of a specific GLib header. Including single
GLib headers is deprecated. Fixes bug #563904.
2008-12-10 08:19:13 +00:00
Julien Moutte
b7f763b23f gst-libs/gst/riff/riff-media.c: Support higher max audio rates for some formats (WAV, Vorbis, LPCM).
Original commit message from CVS:
2008-12-09  Julien Moutte  <julien@fluendo.com>

* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Support higher max audio rates for some formats (WAV, Vorbis, LPCM).
2008-12-09 18:30:10 +00:00
Stefan Kost
b3cc87185a gst-libs/gst/riff/riff-read.c: Fix handling of odd chunks in riff metadata.
Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c:
Fix handling of odd chunks in riff metadata.
2008-12-09 17:21:37 +00:00
Olivier Crete
3c9df39c15 gst-libs/gst/rtp/gstrtcpbuffer.*: Implement gst_rtcp_packet_remove(). Fixes #563174.
Original commit message from CVS:
Patch by: Olivier Crete  <tester at tester ca>
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_remove):
* gst-libs/gst/rtp/gstrtcpbuffer.h:
Implement gst_rtcp_packet_remove(). Fixes #563174.
* tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
Add unit test for some RTCP functions.
2008-12-08 12:08:32 +00:00
이문형
933186aaa1 gst-libs/gst/app/gstappsrc.c: Don't forget to release the lock again if we bail out because some pad is flushing or w...
Original commit message from CVS:
Patch by: 이문형 <iwings at gmail dot com>
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_push_buffer):
Don't forget to release the lock again if we bail out because some
pad is flushing or we've reached EOS, otherwise things will lock up
next time _push_buffer() is called (#562802).
2008-12-01 19:36:35 +00:00
Wim Taymans
af354dbef3 gst-libs/gst/audio/gstbaseaudiosrc.c: Avoid nasty int overflows after about 12 hours and 25 minutes when these code p...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_get_offset), (gst_base_audio_src_create):
Avoid nasty int overflows after about 12 hours and 25 minutes when these
code paths are triggered.
A free beer to Håvard Graff for finding this!
2008-11-27 16:47:41 +00:00
이문형
d80a5c9dbc gst-libs/gst/rtsp/gstrtspconnection.c: A successful gst_poll_wait() doesn't always mean successful connect() on
Original commit message from CVS:
Patch by: 이문형 <iwings at gmail dot com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_connect):
A successful gst_poll_wait() doesn't always mean successful connect() on
Windows.  We should check errors by calling gst_poll_fd_has_error().
See #561924.
2008-11-27 11:16:44 +00:00
Wim Taymans
b2004e3d05 gst-libs/gst/rtp/gstrtcpbuffer.c: Fix typo in the docs.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtcpbuffer.c:
Fix typo in the docs.
2008-11-25 15:33:30 +00:00
Wim Taymans
6983c1c85b gst-libs/gst/audio/gstbaseaudiosink.c: Really fix audiosink drain handling by keeping track of the running_time of th...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain), (gst_base_audio_sink_event),
(gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
(gst_base_audio_sink_change_state):
Really fix audiosink drain handling by keeping track of the running_time
of the last sample.
2008-11-25 10:32:49 +00:00
Stefan Kost
a8264f66c7 gst-libs/gst/audio/gstbaseaudiosink.c: Time is already in running_time. Remove base_time handling. Fixes audiosinks n...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
Time is already in running_time. Remove base_time handling. Fixes
audiosinks not draining and thus chopping some audio in the end.
2008-11-24 20:11:52 +00:00
Stefan Kost
7f937c99d4 gst-libs/gst/audio/gstbaseaudiosink.c: Add one log message to check for audio_drained. Sync one log message with the ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
Add one log message to check for audio_drained. Sync one log message
with the condition. Send EOS after draining audio in pull mode.
2008-11-24 12:56:54 +00:00
Michael Smith
77c3f8bb7f gst-libs/gst/rtsp/gstrtspdefs.c: Fix win32 build. Oops.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspdefs.c:
Fix win32 build. Oops.
2008-11-20 22:06:05 +00:00
Michael Smith
4f04294e45 gst-libs/gst/rtsp/gstrtspdefs.c: Use WSAGetLastError() rather than errno/h_errno on win32.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspdefs.c:
Use WSAGetLastError() rather than errno/h_errno on win32.
2008-11-20 21:40:49 +00:00
Michael Smith
eabff64640 gst-libs/gst/riff/riff-media.c: Support WMA Lossless properly.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
Support WMA Lossless properly.
2008-11-20 21:20:27 +00:00
Jan Schmidt
66ba67723e gst-libs/gst/interfaces/propertyprobe.c: Fix random type causing a docs warning.
Original commit message from CVS:
* gst-libs/gst/interfaces/propertyprobe.c:
Fix random type causing a docs warning.
2008-11-14 21:39:09 +00:00
Wim Taymans
9c32e1f152 gst-libs/gst/rtp/gstrtpbuffer.c: Avoid expensive type checks we already did as part of the _validate() function that ...
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data),
(gst_rtp_buffer_set_packet_len), (gst_rtp_buffer_get_packet_len),
(gst_rtp_buffer_get_header_len), (gst_rtp_buffer_get_version),
(gst_rtp_buffer_set_version), (gst_rtp_buffer_get_padding),
(gst_rtp_buffer_set_padding), (gst_rtp_buffer_pad_to),
(gst_rtp_buffer_get_extension), (gst_rtp_buffer_set_extension),
(gst_rtp_buffer_get_extension_data),
(gst_rtp_buffer_set_extension_data), (gst_rtp_buffer_get_ssrc),
(gst_rtp_buffer_set_ssrc), (gst_rtp_buffer_get_csrc_count),
(gst_rtp_buffer_get_csrc), (gst_rtp_buffer_set_csrc),
(gst_rtp_buffer_get_marker), (gst_rtp_buffer_set_marker),
(gst_rtp_buffer_get_payload_type),
(gst_rtp_buffer_set_payload_type), (gst_rtp_buffer_get_seq),
(gst_rtp_buffer_set_seq), (gst_rtp_buffer_get_timestamp),
(gst_rtp_buffer_set_timestamp),
(gst_rtp_buffer_get_payload_subbuffer),
(gst_rtp_buffer_get_payload_len), (gst_rtp_buffer_get_payload):
Avoid expensive type checks we already did as part of the
_validate() function that should be called first.
2008-11-13 15:37:40 +00:00
Wim Taymans
c98d4a5031 gst-libs/gst/rtp/gstbasertpdepayload.c: Fix some cases where a newsegment event was not sent.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c: (create_segment_event),
(gst_base_rtp_depayload_push_full),
(gst_base_rtp_depayload_set_gst_timestamp):
Fix some cases where a newsegment event was not sent.
2008-11-11 16:40:50 +00:00
Edward Hervey
dfa2705aa0 gst/: Wim, you're a bad boy. You don't want people to contact you or what?
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c:
* gst-libs/gst/app/gstappsrc.c:
* gst/h264parse/gsth264parse.c:
Wim, you're a bad boy. You don't want people to contact you or what?
2008-11-10 14:53:45 +00:00
Wim Taymans
e701e64005 gst-libs/gst/audio/gstbaseaudiosink.c: Use gst_base_sink_do_preroll() to wait for PLAYING and before waiting for the ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
(gst_base_audio_sink_callback):
Use gst_base_sink_do_preroll() to wait for PLAYING and before waiting
for the latency to expire, fixes #559567.
2008-11-10 14:22:09 +00:00
Wim Taymans
67b54151fe gst-libs/gst/app/gstappsrc.*: Add is-live property.
Original commit message from CVS:
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
(gst_app_src_init), (gst_app_src_set_property),
(gst_app_src_get_property), (gst_app_src_push_buffer):
* gst-libs/gst/app/gstappsrc.h:
Add is-live property.
Add some more docs.
2008-11-07 17:35:46 +00:00
Wim Taymans
7ae1871c75 gst-libs/gst/riff/riff-media.c: Fix case where we don't have a range for the rates or channels as is the case with tr...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Fix case where we don't have a range for the rates or channels as is the
case with truespeech.
2008-11-06 12:14:51 +00:00
Stefan Kost
b9d45d9434 Don't install static libs for plugins. Fixes #550851 for -bad.
Original commit message from CVS:
* ext/alsaspdif/Makefile.am:
* ext/amrwb/Makefile.am:
* ext/apexsink/Makefile.am:
* ext/arts/Makefile.am:
* ext/artsd/Makefile.am:
* ext/audiofile/Makefile.am:
* ext/audioresample/Makefile.am:
* ext/bz2/Makefile.am:
* ext/cdaudio/Makefile.am:
* ext/celt/Makefile.am:
* ext/dc1394/Makefile.am:
* ext/dirac/Makefile.am:
* ext/directfb/Makefile.am:
* ext/divx/Makefile.am:
* ext/dts/Makefile.am:
* ext/faac/Makefile.am:
* ext/faad/Makefile.am:
* ext/gsm/Makefile.am:
* ext/hermes/Makefile.am:
* ext/ivorbis/Makefile.am:
* ext/jack/Makefile.am:
* ext/jp2k/Makefile.am:
* ext/ladspa/Makefile.am:
* ext/lcs/Makefile.am:
* ext/libfame/Makefile.am:
* ext/libmms/Makefile.am:
* ext/metadata/Makefile.am:
* ext/mpeg2enc/Makefile.am:
* ext/mplex/Makefile.am:
* ext/musepack/Makefile.am:
* ext/musicbrainz/Makefile.am:
* ext/mythtv/Makefile.am:
* ext/nas/Makefile.am:
* ext/neon/Makefile.am:
* ext/ofa/Makefile.am:
* ext/polyp/Makefile.am:
* ext/resindvd/Makefile.am:
* ext/sdl/Makefile.am:
* ext/shout/Makefile.am:
* ext/snapshot/Makefile.am:
* ext/sndfile/Makefile.am:
* ext/soundtouch/Makefile.am:
* ext/spc/Makefile.am:
* ext/swfdec/Makefile.am:
* ext/tarkin/Makefile.am:
* ext/theora/Makefile.am:
* ext/timidity/Makefile.am:
* ext/twolame/Makefile.am:
* ext/x264/Makefile.am:
* ext/xine/Makefile.am:
* ext/xvid/Makefile.am:
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/dshow/Makefile.am:
* gst/aiffparse/Makefile.am:
* gst/app/Makefile.am:
* gst/audiobuffer/Makefile.am:
* gst/bayer/Makefile.am:
* gst/cdxaparse/Makefile.am:
* gst/chart/Makefile.am:
* gst/colorspace/Makefile.am:
* gst/dccp/Makefile.am:
* gst/deinterlace/Makefile.am:
* gst/deinterlace2/Makefile.am:
* gst/dvdspu/Makefile.am:
* gst/festival/Makefile.am:
* gst/filter/Makefile.am:
* gst/flacparse/Makefile.am:
* gst/flv/Makefile.am:
* gst/games/Makefile.am:
* gst/h264parse/Makefile.am:
* gst/librfb/Makefile.am:
* gst/mixmatrix/Makefile.am:
* gst/modplug/Makefile.am:
* gst/mpeg1sys/Makefile.am:
* gst/mpeg4videoparse/Makefile.am:
* gst/mpegdemux/Makefile.am:
* gst/mpegtsmux/Makefile.am:
* gst/mpegvideoparse/Makefile.am:
* gst/mve/Makefile.am:
* gst/nsf/Makefile.am:
* gst/nuvdemux/Makefile.am:
* gst/overlay/Makefile.am:
* gst/passthrough/Makefile.am:
* gst/pcapparse/Makefile.am:
* gst/playondemand/Makefile.am:
* gst/rawparse/Makefile.am:
* gst/real/Makefile.am:
* gst/rtjpeg/Makefile.am:
* gst/rtpmanager/Makefile.am:
* gst/scaletempo/Makefile.am:
* gst/sdp/Makefile.am:
* gst/selector/Makefile.am:
* gst/smooth/Makefile.am:
* gst/smoothwave/Makefile.am:
* gst/speed/Makefile.am:
* gst/speexresample/Makefile.am:
* gst/stereo/Makefile.am:
* gst/subenc/Makefile.am:
* gst/tta/Makefile.am:
* gst/vbidec/Makefile.am:
* gst/videodrop/Makefile.am:
* gst/videosignal/Makefile.am:
* gst/virtualdub/Makefile.am:
* gst/vmnc/Makefile.am:
* gst/y4m/Makefile.am:
* sys/acmenc/Makefile.am:
* sys/cdrom/Makefile.am:
* sys/dshowdecwrapper/Makefile.am:
* sys/dshowsrcwrapper/Makefile.am:
* sys/dvb/Makefile.am:
* sys/dxr3/Makefile.am:
* sys/fbdev/Makefile.am:
* sys/oss4/Makefile.am:
* sys/qcam/Makefile.am:
* sys/qtwrapper/Makefile.am:
* sys/vcd/Makefile.am:
* sys/wininet/Makefile.am:
* win32/common/config.h:
Don't install static libs for plugins. Fixes #550851 for -bad.
2008-11-04 12:42:18 +00:00
Damien Lespiau
81724500ec gst-libs/gst/rtsp/gstrtspconnection.c: Make the next call to poll not depend on previous calls to poll with or withou...
Original commit message from CVS:
Patch by: Damien Lespiau  <damien.lespiau gmail com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_write):
Make the next call to poll not depend on previous calls to poll with or
without reading from the active descriptor. Fixes #544293.
2008-11-03 10:49:24 +00:00
Nick Haddad
7413c8ee97 gst-libs/gst/riff/: Add support for other fourcc codes that are commonly used for 'uncompressed RGB', including 'RGB ...
Original commit message from CVS:
Patch by: Nick Haddad <nick at haddads dot net>
* gst-libs/gst/riff/riff-ids.h:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
Add support for other fourcc codes that are commonly used for
'uncompressed RGB', including 'RGB ', 'RAW ', and 0.
Fixes #558553.
2008-10-31 09:49:57 +00:00
Wim Taymans
1f724549e4 gst-libs/gst/app/gstappsink.c: Fix the docs.
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init):
Fix the docs.
2008-10-29 17:02:55 +00:00
Sebastian Dröge
4d7ebf29d9 Move float endianness conversion macros to core. Second part of bug ##555196.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/floatcast/floatcast.h:
Move float endianness conversion macros to core. Second part of
bug ##555196.
2008-10-23 07:11:23 +00:00
Sebastian Dröge
4177ce6727 gst-libs/gst/tag/tags.c: Remove useless buffer size assignment. It already has this value.
Original commit message from CVS:
* gst-libs/gst/tag/tags.c: (gst_tag_image_data_to_image_buffer):
Remove useless buffer size assignment. It already has this value.
2008-10-22 12:01:32 +00:00
Wim Taymans
6eed8ca285 gst-libs/gst/audio/gstaudiosink.c: Implement a separate activate functions to start monitoring the segments or, in pu...
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c:
(gst_audioringbuffer_class_init), (gst_audioringbuffer_acquire),
(gst_audioringbuffer_activate), (gst_audioringbuffer_release),
(gst_audioringbuffer_stop):
Implement a separate activate functions to start monitoring the segments
or, in pull mode, pulling in data.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_init), (gst_base_audio_sink_dispose),
(gst_base_audio_sink_query_pad), (gst_base_audio_sink_query),
(gst_base_audio_sink_setcaps), (gst_base_audio_sink_callback),
(gst_base_audio_sink_activate_pull),
(gst_base_audio_sink_async_play),
(gst_base_audio_sink_change_state):
Implement pad and element convert query function.
Activate the ringbuffer.
Use the segment last_stop value as the offset to pull.
Use new basesink _do_preroll() method to preroll in the pulling thread.
Take appropriate locking in the pulling thread.
* gst-libs/gst/audio/gstringbuffer.h:
Update some docs.
2008-10-20 15:35:37 +00:00
Wim Taymans
a6b78893c0 Add methods to more accuratly control the pulling thread of a ringbuffer.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_convert),
(gst_ring_buffer_activate), (gst_ring_buffer_is_active):
* gst-libs/gst/audio/gstringbuffer.h:
Add methods to more accuratly control the pulling thread of a
ringbuffer.
Add format conversion helper code to the ringbuffer.
API: GstRingBuffer:gst_ring_buffer_activate()
API: GstRingBuffer:gst_ring_buffer_is_active()
API: GstRingBuffer:gst_ring_buffer_convert()
2008-10-17 13:19:05 +00:00
Wim Taymans
927999603a gst-libs/gst/audio/gstaudiosink.c: Signal thread startup earlier so that we can immediatly go into pull mode when we ...
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
(gst_audioringbuffer_acquire), (gst_audioringbuffer_release),
(gst_audioringbuffer_stop):
Signal thread startup earlier so that we can immediatly go into pull
mode when we have to and block on preroll.
2008-10-16 15:44:37 +00:00
Wim Taymans
7bd29abb9d gst-libs/gst/audio/gstringbuffer.c: In pull mode we want the callback to prepull a buffer we can preroll on even when...
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_prepare_read):
In pull mode we want the callback to prepull a buffer we can preroll on
even when we are not yet playing.
2008-10-16 15:38:50 +00:00
Edward Hervey
2d1806931d gst-libs/gst/riff/riff-media.c: Add mappping for the KMVC (Karl Morton's Video) Codec.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
(gst_riff_create_video_template_caps):
Add mappping for the KMVC (Karl Morton's Video) Codec.
2008-10-15 15:28:41 +00:00
Wim Taymans
4ae82906ab gst-libs/gst/rtp/gstbasertpdepayload.*: Add some more G_LIKELY
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
(gst_base_rtp_depayload_set_gst_timestamp),
(gst_base_rtp_depayload_change_state):
* gst-libs/gst/rtp/gstbasertpdepayload.h:
Add some more G_LIKELY
Fail when the setcaps function was not called.
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_set_outcaps):
Propagate return value of setcaps.
2008-10-13 09:16:59 +00:00
Sebastian Dröge
796fdbdf17 gst-libs/gst/tag/tags.c: Don't drop the last byte of image tags if they're not an URI list.
Original commit message from CVS:
* gst-libs/gst/tag/tags.c: (gst_tag_image_data_to_image_buffer):
Don't drop the last byte of image tags if they're not an URI list.
Fixes bug #556066.
2008-10-13 08:15:13 +00:00
Edward Hervey
57b0f5bef6 gst-libs/gst/audio/gstbaseaudiosrc.c: Fix debug statements (space between '%' and actual format).
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
Fix debug statements (space between '%' and actual format).
2008-10-08 15:30:33 +00:00
Jan Gerber
76b6a56acb gst-libs/gst/riff/riff-media.c: Add FFV1 fourcc to support playback of FFMPEG lossless video in AVI. Fixes bug #555319.
Original commit message from CVS:
Patch by: Jan Gerber <j at oil21 dot org>
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
(gst_riff_create_video_template_caps):
Add FFV1 fourcc to support playback of FFMPEG lossless video
in AVI. Fixes bug #555319.
2008-10-08 09:22:26 +00:00
Håvard Graff
11086cf6f8 gst-libs/gst/audio/gstbaseaudiosrc.c: Implement skew clock slaving. Fixes #552559.
Original commit message from CVS:
Patch by: Håvard Graff <havard dot graff at tandberg dot com>
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
Implement skew clock slaving. Fixes #552559.
2008-10-08 09:12:36 +00:00
Wim Taymans
dd01a1e56a gst-libs/gst/audio/: Fix include of config.h
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
* gst-libs/gst/audio/testchannels.c:
Fix include of config.h
2008-10-08 09:10:23 +00:00
Tero Saarni
2f96504ff9 gst-libs/gst/sdp/gstsdpmessage.c: Fix parsing of the c= field containing multicast addresses.
Original commit message from CVS:
Based on Patch by: Tero Saarni <tero dot saarni at gmail dot com>
* gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_parse_line),
(print_media), (gst_sdp_message_dump):
Fix parsing of the c= field containing multicast addresses.
Fixes #552199.
Add the connection info to the session or streams.
Fix parsing of the bandwidth.
Add debugging for the connections and bandwidths for a media.
Add debugging for the bandwidth of the session.
2008-10-06 16:36:20 +00:00
Wim Taymans
a2eb053641 gst-libs/gst/rtp/gstbasertppayload.c: Configure the next seqnum and timestamp in the state change so that they can be...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_change_state):
Configure the next seqnum and timestamp in the state change so that they
can be queried soon after.
2008-10-06 16:31:27 +00:00
Wim Taymans
b86ef2dcf2 gst-libs/gst/rtp/gstbasertpdepayload.c: Improve debugging of the rtptime.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_chain):
Improve debugging of the rtptime.
2008-10-06 16:29:33 +00:00
Wim Taymans
c9566ebd68 gst-libs/gst/rtsp/gstrtspmessage.c: Fix the g_return_val_if_fail() statements.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspmessage.c:
(gst_rtsp_message_parse_request),
(gst_rtsp_message_parse_response):
Fix the g_return_val_if_fail() statements.
2008-09-23 17:48:14 +00:00
Michael Smith
46ce5c3737 gst-libs/gst/tag/gsttagdemux.c: Fail to activate if there's insufficient data in the file to be usable, preventing an...
Original commit message from CVS:
* gst-libs/gst/tag/gsttagdemux.c:
Fail to activate if there's insufficient data in the file to be usable,
preventing an assertion fail later. Fixes #552960
2008-09-22 17:44:14 +00:00
Tim-Philipp Müller
029f05635d gst/: Recognise Kate subtitle streams (#550582).
Original commit message from CVS:
* gst-libs/gst/pbutils/descriptions.c:
* gst/typefind/gsttypefindfunctions.c:
Recognise Kate subtitle streams (#550582).
2008-09-15 15:11:18 +00:00
Tim-Philipp Müller
b579580991 gst-libs/gst/audio/audio.h: Remove trailing comma from enum list, which causes problems with -pendantic (#550729).
Original commit message from CVS:
* gst-libs/gst/audio/audio.h: (GST_AUDIO_FIELD_SIGNED):
Remove trailing comma from enum list, which causes problems
with -pendantic (#550729).
2008-09-13 11:04:02 +00:00
Tim-Philipp Müller
f5a176bb6c gst-libs/gst/interfaces/propertyprobe.c: More sanity checks for our second-favourite interface.
Original commit message from CVS:
* gst-libs/gst/interfaces/propertyprobe.c:
(gst_property_probe_get_properties),
(gst_property_probe_get_property),
(gst_property_probe_probe_property),
(gst_property_probe_probe_property_name),
(gst_property_probe_needs_probe),
(gst_property_probe_needs_probe_name),
(gst_property_probe_get_values),
(gst_property_probe_get_values_name),
(gst_property_probe_probe_and_get_values),
(gst_property_probe_probe_and_get_values_name):
More sanity checks for our second-favourite interface.
2008-09-05 19:04:47 +00:00
Stefan Kost
aa7b4f5ac7 gst-libs/gst/interfaces/propertyprobe.c: Check for NULL pointer, in the hope that this fixes #532864.
Original commit message from CVS:
* gst-libs/gst/interfaces/propertyprobe.c:
Check for NULL pointer, in the hope that this fixes #532864.
2008-09-05 14:12:01 +00:00
Wim Taymans
265a494de5 gst-libs/gst/audio/gstaudiosrc.c: Disable a code path that is now called but causes a deadlock for some reason and is...
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_stop):
Disable a code path that is now called but causes a deadlock for some
reason and is unneeded.
2008-09-04 16:25:06 +00:00
Edward Hervey
21952ab09a gst-libs/gst/riff/riff-media.c: Use a decent caps for TrueSpeech instead of a ffmpeg-specific one.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Use a decent caps for TrueSpeech instead of a ffmpeg-specific one.
This will also be fixed for upcoming gst-ffmpeg release so that once
this release of -base is out, it will work with the latest gst-ffmpeg
release.
2008-09-03 14:00:06 +00:00
Edward Hervey
3a0b6ebcfb gst-libs/gst/riff/riff-media.c: Add Truespeech mapping for RIFF formats (AVI/WAV).
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
(gst_riff_create_audio_template_caps):
Add Truespeech mapping for RIFF formats (AVI/WAV).
Fixes #550656
2008-09-03 13:27:20 +00:00
Stefan Kost
54acaa5706 Use new geo location tags from core. Fixes #481169
Original commit message from CVS:
* gst-libs/gst/tag/gstvorbistag.c:
* tests/check/libs/tag.c:
Use new geo location tags from core. Fixes #481169
2008-09-02 06:37:04 +00:00
Wim Taymans
da76d5e7cb gst-libs/gst/audio/gstaudiosink.c: Since we now call stop, we trigger this code path that causes a deadlock is appare...
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_stop):
Since we now call stop, we trigger this code path that causes a deadlock
is apparently not needed.
2008-08-26 17:24:31 +00:00
Wim Taymans
440432612b gst-libs/gst/audio/gstringbuffer.c: Also allow the case where the ringbuffer was paused when we try to stop it so tha...
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_start),
(gst_ring_buffer_stop):
Also allow the case where the ringbuffer was paused when we try to stop
it so that the basesrc stop function is still called.
2008-08-26 15:45:36 +00:00
Sebastian Dröge
12bccedb68 gst-libs/gst/pbutils/missing-plugins.c: When cleaning up the caps fields also remove "depth" for the same reason we r...
Original commit message from CVS:
* gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
When cleaning up the caps fields also remove "depth" for the same
reason we remove "width".
2008-08-15 07:24:38 +00:00
Tim-Philipp Müller
87384c993d gst-libs/gst/pbutils/descriptions.c: Add Lead H.264 here as well.
Original commit message from CVS:
* gst-libs/gst/pbutils/descriptions.c: (format_info_get_desc):
Add Lead H.264 here as well.
2008-08-14 17:14:53 +00:00
Julien Moutte
33690da61a gst-libs/gst/riff/riff-media.c: Add Lead H.264 variant.
Original commit message from CVS:
2008-08-14  Julien Moutte  <julien@fluendo.com>

* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
(gst_riff_create_video_template_caps): Add Lead H.264 variant.
2008-08-14 15:17:31 +00:00
Wim Taymans
510a5befc1 gst-libs/gst/audio/gstbaseaudiosrc.c: When not slaved to another clock also subtract the base_time from our internal ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
When not slaved to another clock also subtract the base_time from our
internal clock time to get the running time.
2008-08-13 09:17:38 +00:00
Stefan Kost
c5ad1c7228 gst-libs/gst/interfaces/propertyprobe.h: Remove double "interface" from doc-string.
Original commit message from CVS:
* gst-libs/gst/interfaces/propertyprobe.h:
Remove double "interface" from doc-string.
* gst-libs/gst/interfaces/xoverlay.h:
Document interface.
* gst-libs/gst/riff/riff.c:
Add basic doc blobs.
2008-08-12 06:31:49 +00:00
Stefan Kost
5d2049cdb3 gst-libs/gst/audio/Makefile.am: Don't try to build that example anymore.
Original commit message from CVS:
* gst-libs/gst/audio/Makefile.am:
Don't try to build that example anymore.
2008-08-11 15:05:35 +00:00
Stefan Kost
3511b2772b gst-libs/gst/audio/: Move audiofiltertemplate to gst-template.
Original commit message from CVS:
* gst-libs/gst/audio/.cvsignore:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
* gst-libs/gst/audio/make_filter:
Move audiofiltertemplate to gst-template.
2008-08-11 14:51:58 +00:00
Stefan Kost
01554ac056 More docs and shuffling. What can we do with the hundreds of #defines.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/gstaudiosrc.h:
More docs and shuffling. What can we do with the hundreds of #defines.
2008-08-11 09:20:33 +00:00
Stefan Kost
f73aa5b817 gst-libs/gst/: Reducing number of dundocumented symbols.
Original commit message from CVS:
* gst-libs/gst/audio/audio.h:
* gst-libs/gst/audio/gstaudiofilter.h:
* gst-libs/gst/audio/gstringbuffer.h:
* gst-libs/gst/interfaces/propertyprobe.h:
* gst-libs/gst/tag/gsttagdemux.h:
Reducing number of dundocumented symbols.
2008-08-11 08:34:56 +00:00
Stefan Kost
26ad0ba982 gst-libs/gst/audio/audio.c: Fix doc comment syntax.
Original commit message from CVS:
* gst-libs/gst/audio/audio.c:
Fix doc comment syntax.
* gst-libs/gst/interfaces/propertyprobe.c:
Add more doc-comments and a FIXME: for the signal.
2008-08-11 07:16:30 +00:00
Frederic Crozat
89be246154 Make sure gettext returns translations in UTF-8 encoding rather than in the current locale encoding (#546822).
Original commit message from CVS:
Patch by: Frederic Crozat <fcrozat@mandriva.org>
* ext/alsa/gstalsaplugin.c: (plugin_init):
* ext/cdparanoia/gstcdparanoiasrc.c: (plugin_init):
* ext/gnomevfs/gstgnomevfs.c: (plugin_init):
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_plugin_init):
* gst-libs/gst/audio/gstbaseaudiosrc.c: (_do_init):
* gst-libs/gst/pbutils/pbutils.c: (gst_pb_utils_init):
* gst-libs/gst/tag/tags.c: (gst_tag_register_tags_internal):
* gst/playback/gstdecodebin.c: (plugin_init):
* gst/playback/gstdecodebin2.c: (gst_decode_bin_plugin_init):
* gst/playback/gstplayback.c: (plugin_init):
* gst/playback/gstqueue2.c: (plugin_init):
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_plugin_init):
* sys/v4l/gstv4l.c: (plugin_init):
Make sure gettext returns translations in UTF-8 encoding rather
than in the current locale encoding (#546822).
2008-08-07 15:58:58 +00:00
Stefan Kost
cac199ce36 gst-libs/gst/pbutils/descriptions.c: Add audio/x-qdm for qtdemux.
Original commit message from CVS:
* gst-libs/gst/pbutils/descriptions.c:
Add audio/x-qdm for qtdemux.
2008-08-06 13:12:07 +00:00
Stefan Kost
f7a085edaa Bump requirement to latest core and use new tag for riff formats.
Original commit message from CVS:
* configure.ac:
* gst-libs/gst/riff/riff-read.c:
Bump requirement to latest core and use new tag for riff formats.
Needed for #520694.
2008-08-01 11:55:07 +00:00
Stefan Kost
4f37ce04f6 gst-libs/gst/riff/riff-ids.h: Adding acid chunk for tempo and loop information.
Original commit message from CVS:
* gst-libs/gst/riff/riff-ids.h:
Adding acid chunk for tempo and loop information.
2008-07-30 09:02:31 +00:00
Damien Lespiau
d76e33616c gst-libs/gst/sdp/gstsdpmessage.c: Use GST_STR_NULL to avoid crashes with libcs that don't like NULL strings in printf...
Original commit message from CVS:
Patch by: Damien Lespiau  <damien.lespiau gmail com>
* gst-libs/gst/sdp/gstsdpmessage.c: (print_media):
Use GST_STR_NULL to avoid crashes with libcs that don't
like NULL strings in printf args (such as the win32 one).
Fixes #544306.
2008-07-23 13:17:31 +00:00
David Schleef
47eafd3466 gst-libs/gst/video/video.c: Revert ABI change.
Original commit message from CVS:
* gst-libs/gst/video/video.c: Revert ABI change.
2008-07-15 22:43:16 +00:00
Sebastian Dröge
dd7d36320e gst-libs/gst/riff/riff-media.c: Make it impossible to have NULL caps at the point where we set framerate and other th...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
Make it impossible to have NULL caps at the point where we set
framerate and other things. Also don't return immediately for "3ivd"
video and let framerate, etc be set. Might fix bug #542508.
2008-07-15 13:05:04 +00:00
Mark Nauwelaerts
d6d5f88174 gst-libs/gst/video/video.c: Video format can also be conveniently determined from (many) non-fixed caps.
Original commit message from CVS:
* gst-libs/gst/video/video.c: (gst_video_format_parse_caps):
Video format can also be conveniently determined from (many)
non-fixed caps.
2008-07-14 17:06:26 +00:00
Damien Lespiau
c3e1de9033 gst-libs/gst/sdp/gstsdpmessage.c: Makes libgstsdp compile with mingw32 by defining the right WINVER so that getaddrin...
Original commit message from CVS:
Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
* gst-libs/gst/sdp/gstsdpmessage.c:
Makes libgstsdp compile with mingw32 by defining the right WINVER so
that getaddrinfo() can be used. Fixes #541358.
2008-07-03 09:12:49 +00:00
Sebastian Dröge
1aca2efee8 gst-libs/gst/riff/: Add support for ADPCM IMA DK3 and DK4 variant in RIFF containers.
Original commit message from CVS:
* gst-libs/gst/riff/riff-ids.h:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
(gst_riff_create_audio_template_caps):
Add support for ADPCM IMA DK3 and DK4 variant in RIFF containers.
Fixes bug #540351.
2008-06-30 09:20:59 +00:00
Sam Morris
752cf09704 gst-libs/gst/interfaces/mixertrack.c: API: Add "index" property to GstMixerTrack to differantiate between multiple mi...
Original commit message from CVS:
Patch by: Sam Morris <sam at robots dot org to uk>
* gst-libs/gst/interfaces/mixertrack.c:
(gst_mixer_track_class_init), (gst_mixer_track_get_property),
(gst_mixer_track_set_property):
API: Add "index" property to GstMixerTrack to differantiate between
multiple mixer tracks with the same label.
* ext/alsa/gstalsamixeroptions.c: (gst_alsa_mixer_options_new):
* ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
Set the "index" property of GstMixerTrack to the index given by ALSA.
Fixes bug #528299.
2008-06-26 06:03:38 +00:00
Wim Taymans
d2f328f55b gst-libs/gst/audio/gstbaseaudiosink.c: Report latency even if we are not live instead of hiding it.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_query), (gst_base_audio_sink_skew_slaving),
(gst_base_audio_sink_render):
Report latency even if we are not live instead of hiding it.
Take ts-offset and render-delay of the basesink into account when
scheduling samples.
Rework the clipping code so that we can take the various offsets into
account and still do correct clipping.
2008-06-20 09:09:37 +00:00
Sebastian Dröge
31f3f65d53 gst-libs/gst/tag/tags.c: Don't increase the size of non-string image buffers by one as this might in theory confuse d...
Original commit message from CVS:
* gst-libs/gst/tag/tags.c: (gst_tag_image_data_to_image_buffer):
Don't increase the size of non-string image buffers by one as this
might in theory confuse decoders. Still increase it by one for string
image buffers to append '\0'.
2008-06-20 08:47:14 +00:00
Andy Wingo
7ba9110637 gst-libs/gst/app/gstappsrc.c (gst_app_src_set_max_bytes)
Original commit message from CVS:
2008-06-16  Andy Wingo  <wingo@pobox.com>

* gst-libs/gst/app/gstappsrc.c (gst_app_src_set_max_bytes)
(gst_app_src_get_max_bytes, gst_app_src_push_buffer): Use
G_GUINT64_FORMAT. Avoid overflow in get_max_bytes().
2008-06-16 14:11:36 +00:00
Stefan Kost
4ad8ad3db5 docs/plugins/: docs/plugins/inspect/plugin-mythtv.xml
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/gst-plugins-bad-plugins.args:
* docs/plugins/gst-plugins-bad-plugins.hierarchy:
* docs/plugins/gst-plugins-bad-plugins.interfaces:
* docs/plugins/gst-plugins-bad-plugins.prerequisites:
* docs/plugins/gst-plugins-bad-plugins.signals:
* docs/plugins/inspect/plugin-alsaspdif.xml:
* docs/plugins/inspect/plugin-amrwb.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-bayer.xml:
* docs/plugins/inspect/plugin-bz2.xml:
* docs/plugins/inspect/plugin-cdaudio.xml:
* docs/plugins/inspect/plugin-cdxaparse.xml:
* docs/plugins/inspect/plugin-dtsdec.xml:
* docs/plugins/inspect/plugin-dvb.xml:
* docs/plugins/inspect/plugin-dvdspu.xml:
* docs/plugins/inspect/plugin-faac.xml:
* docs/plugins/inspect/plugin-faad.xml:
* docs/plugins/inspect/plugin-fbdevsink.xml:
* docs/plugins/inspect/plugin-festival.xml:
* docs/plugins/inspect/plugin-filter.xml:
* docs/plugins/inspect/plugin-flvdemux.xml:
* docs/plugins/inspect/plugin-freeze.xml:
* docs/plugins/inspect/plugin-gsm.xml:
* docs/plugins/inspect/plugin-gstinterlace.xml:
* docs/plugins/inspect/plugin-gstrtpmanager.xml:
* docs/plugins/inspect/plugin-h264parse.xml:
* docs/plugins/inspect/plugin-interleave.xml:
* docs/plugins/inspect/plugin-jack.xml:
* docs/plugins/inspect/plugin-ladspa.xml:
* docs/plugins/inspect/plugin-metadata.xml:
* docs/plugins/inspect/plugin-mms.xml:
* docs/plugins/inspect/plugin-modplug.xml:
* docs/plugins/inspect/plugin-mpeg2enc.xml:
* docs/plugins/inspect/plugin-mpeg4videoparse.xml:
* docs/plugins/inspect/plugin-mpegtsparse.xml:
* docs/plugins/inspect/plugin-mpegvideoparse.xml:
* docs/plugins/inspect/plugin-musepack.xml:
* docs/plugins/inspect/plugin-musicbrainz.xml:
* docs/plugins/inspect/plugin-mve.xml:
* docs/plugins/inspect/plugin-mythtv.xml
* docs/plugins/inspect/plugin-nas.xml:
* docs/plugins/inspect/plugin-neon.xml:
* docs/plugins/inspect/plugin-nsfdec.xml:
* docs/plugins/inspect/plugin-nuvdemux.xml:
* docs/plugins/inspect/plugin-oss4.xml
* docs/plugins/inspect/plugin-rawparse.xml:
* docs/plugins/inspect/plugin-real.xml:
* docs/plugins/inspect/plugin-replaygain.xml:
* docs/plugins/inspect/plugin-rfbsrc.xml:
* docs/plugins/inspect/plugin-sdl.xml:
* docs/plugins/inspect/plugin-sdp.xml:
* docs/plugins/inspect/plugin-selector.xml:
* docs/plugins/inspect/plugin-sndfile.xml:
* docs/plugins/inspect/plugin-soundtouch.xml:
* docs/plugins/inspect/plugin-spcdec.xml:
* docs/plugins/inspect/plugin-speed.xml:
* docs/plugins/inspect/plugin-speexresample.xml:
* docs/plugins/inspect/plugin-stereo.xml:
* docs/plugins/inspect/plugin-subenc.xml
* docs/plugins/inspect/plugin-timidity.xml:
* docs/plugins/inspect/plugin-tta.xml:
* docs/plugins/inspect/plugin-vcdsrc.xml:
* docs/plugins/inspect/plugin-videosignal.xml:
* docs/plugins/inspect/plugin-vmnc.xml:
* docs/plugins/inspect/plugin-wildmidi.xml:
* docs/plugins/inspect/plugin-x264.xml:
* docs/plugins/inspect/plugin-xvid.xml:
* docs/plugins/inspect/plugin-y4menc.xml:
* ext/amrwb/gstamrwbdec.c:
* ext/amrwb/gstamrwbenc.c:
* ext/amrwb/gstamrwbparse.c:
* ext/dc1394/gstdc1394.c:
* ext/directfb/dfbvideosink.c:
* ext/ivorbis/vorbisdec.c:
* ext/jack/gstjackaudiosink.c:
* ext/mpeg2enc/gstmpeg2enc.cc:
* ext/mplex/gstmplex.cc:
* ext/musicbrainz/gsttrm.c:
* ext/mythtv/gstmythtvsrc.c:
* ext/theora/theoradec.c:
* ext/timidity/gsttimidity.c:
* ext/timidity/gstwildmidi.c:
* gst-libs/gst/app/gstappsink.c:
* gst/deinterlace/gstdeinterlace.c:
* gst/dvdspu/gstdvdspu.c:
* gst/festival/gstfestival.c:
* gst/freeze/gstfreeze.c:
* gst/interleave/deinterleave.c:
* gst/interleave/interleave.c:
* gst/modplug/gstmodplug.cc:
* gst/nuvdemux/gstnuvdemux.c:
Add missing elements to docs. Fix doc-markup: use convinience syntax
for examples (produces valid docbook), add several refsec2 when we
have several titles. Fix some types.
2008-06-13 11:59:21 +00:00
Wim Taymans
c30d479783 examples/app/: Add beefed up example app from bug #413418. It now also uses appsink instead of fakesink for more ulti...
Original commit message from CVS:
* examples/app/.cvsignore:
* examples/app/Makefile.am:
* examples/app/appsink-src.c: (on_new_buffer_from_source),
(on_source_message), (on_sink_message), (main):
Add beefed up example app from bug #413418. It now also uses appsink
instead of fakesink for more ultimate coolness.
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
(gst_app_src_init), (gst_app_src_set_property),
(gst_app_src_get_property), (gst_app_src_unlock),
(gst_app_src_unlock_stop), (gst_app_src_create),
(gst_app_src_set_max_bytes), (gst_app_src_push_buffer),
(gst_app_src_end_of_stream):
* gst-libs/gst/app/gstappsrc.h:
Add block property to allow push based implementation to block when we
fill up the appsrc queues.
Emit the enough-data signal while releasing our lock.
2008-06-12 15:47:03 +00:00
Stefan Kost
e54b324d7d Do not use short_description in section docs for elements. We extract them from element details and there will be war...
Original commit message from CVS:
* ext/dc1394/gstdc1394.c:
* ext/ivorbis/vorbisdec.c:
* ext/jack/gstjackaudiosink.c:
* ext/metadata/gstmetadatademux.c:
* ext/mythtv/gstmythtvsrc.c:
* ext/theora/theoradec.c:
* gst-libs/gst/app/gstappsink.c:
* gst/bayer/gstbayer2rgb.c:
* gst/deinterlace/gstdeinterlace.c:
* gst/rawparse/gstaudioparse.c:
* gst/rawparse/gstvideoparse.c:
* gst/rtpmanager/gstrtpbin.c:
* gst/rtpmanager/gstrtpclient.c:
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/rtpmanager/gstrtpptdemux.c:
* gst/rtpmanager/gstrtpsession.c:
* gst/rtpmanager/gstrtpssrcdemux.c:
* gst/selector/gstinputselector.c:
* gst/selector/gstoutputselector.c:
* gst/videosignal/gstvideoanalyse.c:
* gst/videosignal/gstvideodetect.c:
* gst/videosignal/gstvideomark.c:
* sys/oss4/oss4-mixer.c:
* sys/oss4/oss4-sink.c:
* sys/oss4/oss4-source.c:
Do not use short_description in section docs for elements. We extract
them from element details and there will be warnings if they differ.
Also fixing up the ChangeLog order.
2008-06-12 14:49:15 +00:00
David Schleef
526b2e63a2 gst-libs/gst/rtsp/gstrtspconnection.c: Fix build on win32.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspconnection.c:
Fix build on win32.
Patch By: David Schleef <ds@schleef.org>
Fixes: #536874
2008-06-11 20:13:00 +00:00
Wim Taymans
593d4b1af3 examples/app/: Added 3 more example application for using appsrc in random-access mode, pull-mode streaming and pull ...
Original commit message from CVS:
* examples/app/Makefile.am:
* examples/app/appsrc-ra.c: (feed_data), (seek_data),
(found_source), (bus_message), (main):
* examples/app/appsrc-seekable.c: (feed_data), (seek_data),
(found_source), (bus_message), (main):
* examples/app/appsrc-stream2.c: (feed_data), (found_source),
(bus_message), (main):
Added 3 more example application for using appsrc in random-access mode,
pull-mode streaming and pull mode seekable.
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
(gst_app_src_start), (gst_app_src_do_get_size),
(gst_app_src_create):
* gst-libs/gst/app/gstappsrc.h:
Make stream-type property writable.
Unset flushing when starting so that we reuse appsrc.
Inform basesrc about the configured size.
Emit seek-data signal when we are going to a different offset in
random-access mode.
2008-06-06 16:50:51 +00:00
Wim Taymans
20d64607d4 examples/app/: Added an example on how to use appsrc in playbin in streaming mode from an mmapped file.
Original commit message from CVS:
* examples/app/.cvsignore:
* examples/app/Makefile.am:
* examples/app/appsrc-stream.c: (read_data), (start_feed),
(stop_feed), (found_source), (bus_message), (main):
Added an example on how to use appsrc in playbin in streaming mode from
an mmapped file.
* examples/app/appsrc_ex.c: (main):
Set pipeline to NULL to free queued buffers.
* gst-libs/gst/app/gstapp-marshal.list:
* gst-libs/gst/app/gstappsrc.c: (stream_type_get_type), (_do_init),
(gst_app_src_class_init), (gst_app_src_init),
(gst_app_src_flush_queued), (gst_app_src_dispose),
(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_do_seek),
(gst_app_src_create), (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_push_buffer),
(gst_app_src_end_of_stream), (gst_app_src_uri_get_type),
(gst_app_src_uri_get_protocols), (gst_app_src_uri_get_uri),
(gst_app_src_uri_set_uri), (gst_app_src_uri_handler_init):
* gst-libs/gst/app/gstappsrc.h:
Measure max queue size in bytes instead.
Add support for 3 modes of operation, streaming, seekable and
random-access, making basesrc handle the scheduling modes for each.
Add appsrc:// uri handler so that automatic plugging can be done from
playbin2 or uridecodebin, for example.
Added support for custom segment formats.
Add support for push and pull based operations from the application.
Expand the methods so that errors can be detected.
Flush the queued buffers on seeks and when shutting down.
Add signals to inform the app that a seek must happen.
2008-06-05 16:38:50 +00:00
Peter Kjellerstedt
26cd5ea1c8 gst-libs/gst/rtsp/gstrtspconnection.*: Add a couple of missing argument guards.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_set_auth), (gst_rtsp_connection_set_auth_param),
(gst_rtsp_connection_clear_auth_params),
(gst_rtsp_connection_set_qos_dscp), (gst_rtsp_connection_get_ip):
* gst-libs/gst/rtsp/gstrtspconnection.h:
Add a couple of missing argument guards.
Add a way of setting the DSCP for an RTSP connection.
Add an accessor method for the ip member of GstRTSPConnection as all
members are supposed to be private.
2008-06-04 11:53:53 +00:00
Tim-Philipp Müller
44e087f8c9 gst-libs/gst/interfaces/mixertrack.h: Document mixer track flags.
Original commit message from CVS:
* gst-libs/gst/interfaces/mixertrack.h:
Document mixer track flags.
2008-06-04 10:18:42 +00:00
John Millikin
f934d1c233 gst-libs/gst/tag/gstvorbistag.c: Retrieve COVERART tags from vorbis comments (#512333)
Original commit message from CVS:
Based on patch by: John Millikin <jmillikin gmail com>
* gst-libs/gst/tag/gstvorbistag.c: (tag_matches), (gst_vorbis_tag_add),
(gst_vorbis_tag_add_coverart):
Retrieve COVERART tags from vorbis comments (#512333)
2008-06-03 20:01:58 +00:00
Tim-Philipp Müller
8b491df810 gst-libs/gst/tag/: Don't forget to add new enum value here too (should probably use glib-mkenums here...).
Original commit message from CVS:
* gst-libs/gst/tag/tag.h:
* gst-libs/gst/tag/tags.c: (register_tag_image_type_enum):
Don't forget to add new enum value here too (should probably use
glib-mkenums here...).
2008-06-03 19:44:48 +00:00
Tim-Philipp Müller
cd9bb9a674 gst-libs/gst/tag/: API: add gst_tag_image_data_to_image_buffer()
Original commit message from CVS:
* gst-libs/gst/tag/gstid3tag.c: (gst_tag_list_add_id3_image):
* gst-libs/gst/tag/tag.h: (GST_TAG_IMAGE_TYPE_NONE),
* gst-libs/gst/tag/tags.c: (register_tag_image_type_enum),
(gst_tag_image_type_get_type), (gst_tag_image_type_is_valid),
(gst_tag_image_data_to_image_buffer):
Add two utility functions to avoid code duplication (#512333):
API: add gst_tag_image_data_to_image_buffer()
API: add gst_tag_list_add_id3_image()
2008-06-03 19:29:06 +00:00
Sebastian Dröge
0de81029c8 API: Make gst_audio_check_channel_positions() public.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions):
* gst-libs/gst/audio/multichannel.h:
API: Make gst_audio_check_channel_positions() public.
* tests/check/libs/audio.c: (GST_START_TEST):
Add some simple checks for gst_audio_check_channel_positions().
2008-06-03 08:48:32 +00:00
Mark Nauwelaerts
9fa61c528d gst-libs/gst/audio/gstbaseaudiosrc.c: Add a gtk-doc chunk for the new properties to have a Since: indication.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init):
Add a gtk-doc chunk for the new properties to have a Since: indication.
2008-05-31 19:57:57 +00:00
Mark Nauwelaerts
c660bbd6dd gst-libs/gst/audio/gstbaseaudiosrc.c: Provide readable actual-buffer-time and actual-latency-time properties that ref...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init), (gst_base_audio_src_dispose),
(gst_base_audio_src_get_property), (gst_base_audio_src_setcaps),
(gst_base_audio_src_change_state):
Provide readable actual-buffer-time and actual-latency-time properties
that reflect the configured ringbuffer values. Fixes #524724.
2008-05-31 18:10:47 +00:00
Wim Taymans
11309247f3 gst-libs/gst/rtp/gstbasertppayload.c: Simply converting the running time into an RTP timestamp by scaling it based on...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push),
(gst_basertppayload_change_state):
Simply converting the running time into an RTP timestamp by scaling it
based on the clock-rate is good enough for making an RTP timestamp. This
has the added benefit that we can later on expose a property with the
RTP timestamp of running time 0, as is needed for RTSP servers to
generate the response of the PLAY request.
2008-05-30 15:29:20 +00:00
Sebastian Dröge
45ef6b5e13 gst-libs/gst/audio/multichannel.c: Allow rear center together with rear left/right and other previously conflicting c...
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions),
(gst_audio_set_structure_channel_positions_list),
(gst_audio_fixate_channel_positions):
Allow rear center together with rear left/right and other previously
conflicting channel positions. The reason why they weren't allowed
was the channel mixing implementation in audioconvert.
Also take this into account when fixing channel layouts.
Allow setting channel positions for 1/2 channels when using
gst_audio_set_structure_channel_position().
* gst/audioconvert/gstchannelmix.c:
(gst_channel_mix_fill_compatible), (gst_channel_mix_detect_pos),
(gst_channel_mix_fill_one_other), (gst_channel_mix_fill_others),
(gst_channel_mix_fill_special), (gst_channel_mix_fill_matrix):
Major rewrite of the channel mixing.
We now allow previously	conflicting channel positions to appear
together (rear center and rear left/right for example).
Fixes bug #533817.
Rework the way channels are mixed together to take more possible
channel positions into account, properly mix from/to side channels
and don't assume that either center, left&right or nothing of a
specific position is available anymore.
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Adjust unit tests with non-standard 1/2 channel layouts to the more
correct new behaviour.
Add a unit test for 5.1->Stereo downmixing.
2008-05-29 11:34:09 +00:00
Wim Taymans
2855fb48ad gst-libs/gst/cdda/gstcddabasesrc.c: Fix EOS condition and track addition check, the track.end sector is included in t...
Original commit message from CVS:
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_add_track), (gst_cdda_base_src_create):
Fix EOS condition and track addition check, the track.end sector is
included in the track. Fixes #533265.
2008-05-28 15:48:33 +00:00
Wim Taymans
35e4b75b86 gst-libs/gst/audio/gstaudioclock.*: Add method to inform the clock that the time starts from 0 again. We use this inf...
Original commit message from CVS:
* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
(gst_audio_clock_reset), (gst_audio_clock_get_internal_time):
* gst-libs/gst/audio/gstaudioclock.h:
Add method to inform the clock that the time starts from 0 again. We use
this info to calculate a clock offset so that the time we report in
internal_time is monotonically increasing, as required by the clock base
class. Fixes #521761.
API: GstAudioClock::gst_audio_clock_reset()
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_skew_slaving),
(gst_base_audio_sink_change_state):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create), (gst_base_audio_src_change_state):
Reset reported time when we (re)create the ringbuffer.
2008-05-27 16:20:17 +00:00
Wim Taymans
13d7048f69 gst-libs/gst/riff/riff-media.c: Add support for DVCPRO.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
(gst_riff_create_video_template_caps):
Add support for DVCPRO.
2008-05-26 17:18:52 +00:00
Wim Taymans
79a725148d gst-libs/gst/rtp/gstbasertpdepayload.c: Check sequence numbers, mark input buffers with a discont flag for the subcla...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_chain),
(gst_base_rtp_depayload_handle_sink_event),
(gst_base_rtp_depayload_push_full),
(gst_base_rtp_depayload_change_state):
Check sequence numbers, mark input buffers with a discont flag for the
subclass when we detected a gap, drop duplicate buffers. We do this
because one can use the element without a jitterbuffer in front and we
don't want to feed the subclasses invalid or reordered data.
Do an error when the subclass did not provide a process function instead
of crashing.
Some other small cleanups.
2008-05-23 14:14:28 +00:00
Felipe Contreras
b5f896dad6 gst-libs/gst/rtsp/Makefile.am: Distribute, don't install md5.h
Original commit message from CVS:
* gst-libs/gst/rtsp/Makefile.am:
Distribute, don't install md5.h
2008-05-21 17:01:16 +00:00
Stefan Kost
eda6d89b8c gst-libs/gst/riff/riff-media.c: Fix wrong method name in docs. Fix calculation of strf fields for broken mulaw/alaw.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
Fix wrong method name in docs. Fix calculation of strf fields for
broken mulaw/alaw.
* gst-libs/gst/riff/riff-read.c:
Whitespace fix and removing double ';'.
2008-05-21 14:35:41 +00:00
Sebastian Dröge
d03bbd1e3e gst-libs/gst/audio/multichannel.c: Allow non-standard 2 channel layouts.
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions):
Allow non-standard 2 channel layouts.
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Add some tests for converting and remapping non-standard 1 and 2
channel layouts.
2008-05-21 07:39:56 +00:00
Sebastian Dröge
3ee2676c2e gst-libs/gst/rtsp/gstrtsptransport.h: Make the GstRTSPTransport struct members public as there are no setters/getters...
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtsptransport.h:
Make the GstRTSPTransport struct members public as there are no
setters/getters and it's supposed to be changed directly.
Fixes bug #533087.
2008-05-21 06:39:20 +00:00
Wim Taymans
f36d9d6b08 gst-libs/gst/audio/gstbaseaudiosink.c: We can only use our optimal calibration if we prerolled before the latency exp...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_sync_latency):
We can only use our optimal calibration if we prerolled before the
latency expired.
2008-05-20 16:26:53 +00:00
Wim Taymans
d8dc371c0d ext/gnomevfs/gstgnomevfssrc.*: Set the ICY caps on the srcpad from where they get picked up by the base class now and...
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_init),
(gst_gnome_vfs_src_finalize),
(gst_gnome_vfs_src_received_headers_callback),
(gst_gnome_vfs_src_create), (gst_gnome_vfs_src_stop):
* ext/gnomevfs/gstgnomevfssrc.h:
Set the ICY caps on the srcpad from where they get picked up by the base
class now and set on the outgoing buffers.
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
* sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_new):
BaseSrc now sets the caps on outgoing buffers automatically.
2008-05-20 11:13:27 +00:00
Wim Taymans
95d162fb71 gst-libs/gst/audio/gstbaseaudiosink.c: Change the way in which the ringbuffer is started when dealing with a slaved c...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_resample_slaving),
(gst_base_audio_sink_skew_slaving),
(gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
(gst_base_audio_sink_async_play),
(gst_base_audio_sink_change_state):
Change the way in which the ringbuffer is started when dealing with a
slaved clock and latency. We now sync to the clock until we reach
upstream latency before starting the ringbuffer. This has the effect
that we can accurately align the master and slave clocks and let the
rate correction code take care of the initial drift or rounding errors
instead of leaving them uncorrected with the old approach.
2008-05-20 11:09:06 +00:00
Wim Taymans
86ab51207b gst-libs/gst/rtp/gstbasertpdepayload.c: Validate the RTP packet before further processing it. It's just too dangerous...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_chain):
Validate the RTP packet before further processing it. It's just too
dangerous to accept random packets and people are not forced to use a
jitterbuffer or session manager to filter out the bad packets.
* gst-libs/gst/rtp/gstrtpbuffer.c:
(gst_rtp_buffer_set_extension_data),
(gst_rtp_buffer_get_payload_subbuffer):
Small cleanups.
When setting extension data in a buffer that is too small, we fail and
we should not set the extension bit.
Change GST_WARNINGS into g_warning because they really are
programming errors.
* tests/check/libs/rtp.c: (GST_START_TEST):
Catch the g_warnings now in the unit tests and that fact that failing to
set extension data left the extension bit untouched.
2008-05-14 20:28:02 +00:00
Bernard B
d06df554a9 gst-libs/gst/rtp/gstrtpbuffer.c: Fix seqnum compare function for bordercase values and fix the docs again. Fixes #533...
Original commit message from CVS:
Patch by: Bernard B <b-gnome at largestprime dot net>
* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_compare_seqnum):
Fix seqnum compare function for bordercase values and fix the docs
again. Fixes #533075.
* tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
Add a testcase for seqnum compare function.
2008-05-14 13:43:12 +00:00
Wim Taymans
0c9b13988c gst-libs/gst/audio/gstbaseaudiosink.c: Revert previous patch that attempted to more accurately calculate the initial ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render), (gst_base_audio_sink_async_play):
Revert previous patch that attempted to more accurately calculate the
initial offset between master and slave clock. The best thing we can do
in general is take the time of both clocks as the diff since we don't
know when the actual preroll happened.
2008-05-12 08:45:11 +00:00
Tim-Philipp Müller
1482332184 gst-libs/gst/pbutils/install-plugins.c: Fix docs: type and missing word.
Original commit message from CVS:
* gst-libs/gst/pbutils/install-plugins.c:
Fix docs: type and missing word.
2008-05-11 19:52:59 +00:00
Jan Schmidt
f11cf32c3f Document the GstTuner and GstColorBalance interfaces, and some other random API functions that needed it. 70% symbol ...
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/interfaces/colorbalance.c:
* gst-libs/gst/interfaces/colorbalance.h:
* gst-libs/gst/interfaces/colorbalancechannel.c:
* gst-libs/gst/interfaces/colorbalancechannel.h:
* gst-libs/gst/interfaces/tuner.c:
* gst-libs/gst/interfaces/tunerchannel.c:
* gst-libs/gst/interfaces/tunerchannel.h:
* gst-libs/gst/interfaces/tunernorm.c:
* gst-libs/gst/interfaces/tunernorm.h:
* gst-libs/gst/video/video.c:
* gst-libs/gst/video/video.h:
Document the GstTuner and GstColorBalance interfaces, and some
other random API functions that needed it. 70% symbol coverage, woo.
2008-05-09 21:42:26 +00:00
Wim Taymans
fc523e047c gst-libs/gst/audio/gstaudiosink.c: Choose to allocate one less segment but require one additional segment as latency.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_acquire):
Choose to allocate one less segment but require one additional segment
as latency.
* gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_acquire):
No need to increment the number of segments in the source.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_get_time), (clock_convert_external),
(gst_base_audio_sink_resample_slaving),
(gst_base_audio_sink_skew_slaving),
(gst_base_audio_sink_none_slaving), (gst_base_audio_sink_render),
(gst_base_audio_sink_async_play):
Remove adding latency when returning the internal time while subtracting
it again when we use the value a little later.
When calculating the end timestamp, we are making a rounding error
with the current algorithm. Ensure that we don't accumulate these
rounding errors when aligning samples by not resampling at all if we
don't need to. Fixes #419351.
Make the initial calibration of the clock slaving a little more
predictable and accurate. Also handle the case where we don't do
clock slaving.
2008-05-09 16:38:10 +00:00
Wouter Cloetens
a8a2b9c717 gst-libs/gst/rtsp/: Add Digest authorization support for RTSP connections. See #532065.
Original commit message from CVS:
Patch by: Wouter Cloetens <zombie at e2big dot org>
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_create), (md5_digest_to_hex_string),
(auth_digest_compute_hex_urp), (auth_digest_compute_response),
(add_auth_header), (gst_rtsp_connection_free),
(gst_rtsp_connection_set_auth), (str_case_hash), (str_case_equal),
(gst_rtsp_connection_set_auth_param),
(gst_rtsp_connection_clear_auth_params):
* gst-libs/gst/rtsp/gstrtspconnection.h:
Add Digest authorization support for RTSP connections. See #532065.
* gst-libs/gst/rtsp/md5.c:
* gst-libs/gst/rtsp/md5.h:
Yeap, another md5 implementation until we can depend on a glib that has
support for it.
2008-05-08 14:46:27 +00:00
Wim Taymans
09f7dee84d gst-libs/gst/audio/gstbaseaudiosink.c: Report the latency with the new seglatency parameter.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_query):
Report the latency with the new seglatency parameter.
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_debug_spec_buff), (gst_ring_buffer_parse_caps),
(gst_ring_buffer_acquire):
* gst-libs/gst/audio/gstringbuffer.h:
Add new field to the ringbufferspec to specify the expected latency
between the underlying device read/write pointer, this is needed
when writing sinks that sit a little closer to the hardware.
Add some more docs for other fields.
2008-05-07 15:47:03 +00:00
Wim Taymans
8c3b00be4d gst-libs/gst/app/: Add marshal.list, make it compile and add to cvsignore.
Original commit message from CVS:
* gst-libs/gst/app/.cvsignore:
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/app/gstapp-marshal.list:
Add marshal.list, make it compile and add to cvsignore.
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_dispose),
(gst_app_sink_stop):
Small cleanups.
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
(gst_app_src_init), (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_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_seekable),
(gst_app_src_get_seekable), (gst_app_src_set_max_buffers),
(gst_app_src_get_max_buffers), (gst_app_src_push_buffer),
(gst_app_src_end_of_stream):
* gst-libs/gst/app/gstappsrc.h:
Beat appsrc in shape, add signals and actions.
Add some docs.
Add properties for caps, size, seekability and max-buffers.
Fix unlock/stop code.
2008-05-07 10:38:23 +00:00
Wim Taymans
1275eda108 gst-libs/gst/app/gstappsink.*: Start some docs.
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init),
(gst_app_sink_init), (gst_app_sink_set_property),
(gst_app_sink_get_property), (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_set_caps), (gst_app_sink_set_drop),
(gst_app_sink_get_drop):
* gst-libs/gst/app/gstappsink.h:
Start some docs.
Add property to drop buffers when the queue is filled
Fix unlocking and flushing when the queues are filled.
2008-05-05 10:27:45 +00:00
Edward Hervey
b98072f957 gst-libs/gst/pbutils/descriptions.c: It's SorensOn and not SorensEn.
Original commit message from CVS:
* gst-libs/gst/pbutils/descriptions.c: (formats):
It's SorensOn and not SorensEn.
2008-05-05 07:41:03 +00:00
Tim-Philipp Müller
6451cbf5b7 gst-libs/gst/pbutils/descriptions.c: Fix description of video/x-flash-video.
Original commit message from CVS:
* gst-libs/gst/pbutils/descriptions.c: (formats):
Fix description of video/x-flash-video.
2008-05-04 15:23:36 +00:00
Sebastian Dröge
83f0729394 Remove some unused code.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
* gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
* gst/tcp/gsttcp.c: (gst_tcp_socket_write):
* sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_get_fps_list):
Remove some unused code.
* gst/audioconvert/gstaudioquantize.c:
(gst_audio_quantize_free_noise_shaping):
Don't return before freeing the noise shaping history.
2008-05-04 15:02:20 +00:00
Wim Taymans
c6389eec57 gst-libs/gst/rtp/gstbasertppayload.c: Rename the setcaps/getcaps function internally to make it clear that they are c...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
(gst_basertppayload_sink_setcaps),
(gst_basertppayload_sink_getcaps):
Rename the setcaps/getcaps function internally to make it clear that
they are called for the sink pad.
2008-05-02 12:13:08 +00:00
Wim Taymans
f0f6476aff gst-libs/gst/rtp/gstbasertpdepayload.*: Catch packet-lost events from the jitterbuffer and convert them into a vmetho...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_class_init),
(gst_base_rtp_depayload_handle_sink_event), (create_segment_event),
(gst_base_rtp_depayload_packet_lost),
(gst_base_rtp_depayload_set_gst_timestamp):
* gst-libs/gst/rtp/gstbasertpdepayload.h:
Catch packet-lost events from the jitterbuffer and convert them into a
vmethod call (lost-packet) so that depayloaders can do something smart.
Also add a default packet-lost function that sends out a segment update
to the decoders.
2008-05-02 12:11:07 +00:00
Wim Taymans
7916e386ca gst-libs/gst/audio/gstbaseaudiosink.h: Clarify some docs.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.h:
Clarify some docs.
* gst-libs/gst/audio/gstbaseaudiosrc.c: (slave_method_get_type),
(gst_base_audio_src_class_init), (gst_base_audio_src_init),
(gst_base_audio_src_set_slave_method),
(gst_base_audio_src_get_slave_method),
(gst_base_audio_src_set_property),
(gst_base_audio_src_get_property), (gst_base_audio_src_create):
* gst-libs/gst/audio/gstbaseaudiosrc.h:
Add property and methods for selecting the clock slave method in the
source, like in the sink.
We only implement "none" and "re-timestamp" for now.
API: gst_base_audio_src_set_slave_method()
API: gst_base_audio_src_get_slave_method()
2008-04-28 08:51:38 +00:00
Wim Taymans
0c5d01b59f gst-libs/gst/app/gstappsink.*: Add more docs.
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init),
(gst_app_sink_init), (gst_app_sink_set_property),
(gst_app_sink_get_property), (gst_app_sink_event),
(gst_app_sink_preroll), (gst_app_sink_render),
(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_pull_buffer):
* gst-libs/gst/app/gstappsink.h:
Add more docs.
Add signals for when preroll and render buffers are available.
Add property to control signal emission.
Add property to control the max queue size.
2008-04-25 18:18:47 +00:00
Wim Taymans
5b8afead80 gst-libs/gst/rtp/gstrtpbuffer.c: Fix the docs about the seqnum compare function, it returns a difference.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtpbuffer.c:
Fix the docs about the seqnum compare function, it returns a difference.
2008-04-25 07:37:09 +00:00
Sebastian Dröge
b8d03daeab gst-libs/gst/rtp/gstrtppayloads.c: Use g_ascii_strcasecmp() instead of the deprecated g_strcasecmp().
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtppayloads.c:
(gst_rtp_payload_info_for_name):
Use g_ascii_strcasecmp() instead of the deprecated g_strcasecmp().
2008-04-19 16:33:24 +00:00
Sebastian Dröge
66bbadadd0 gst-libs/gst/audio/gstringbuffer.c: Use g_atomic_int_set() instead of gst_atomic_int_set().
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_may_start):
Use g_atomic_int_set() instead of gst_atomic_int_set().
2008-04-17 07:33:46 +00:00
Tim-Philipp Müller
96a3780816 gst-libs/gst/video/video.c: Add guards to these functions to ensure sane input values.
Original commit message from CVS:
* gst-libs/gst/video/video.c: (gst_video_format_new_caps),
(gst_video_format_to_fourcc), (gst_video_format_get_row_stride),
(gst_video_format_get_pixel_stride),
(gst_video_format_get_component_width),
(gst_video_format_get_component_height),
(gst_video_format_get_component_offset), (gst_video_format_get_size),
(gst_video_format_convert):
Add guards to these functions to ensure sane input values.
* tests/check/libs/video.c:
Fix unit test not to create caps with width=0 and height=0.
2008-04-11 17:13:52 +00:00
Tim-Philipp Müller
7a29d716bd gst/: Work around missing bits of thread-safety on older GLibs some more to avoid assertions when starting up multipl...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init):
* gst/playback/gstplayback.c: (plugin_init):
* gst/volume/gstvolume.c: (plugin_init):
Work around missing bits of thread-safety on older GLibs some
more to avoid assertions when starting up multiple playbin
objects concurrently (see #512382).
2008-04-06 20:16:27 +00:00
Tim-Philipp Müller
c1a32ecf46 gst-libs/gst/pbutils/missing-plugins.c: Remove some more fields.
Original commit message from CVS:
* gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
Remove some more fields.
2008-04-06 17:19:39 +00:00
Wim Taymans
ce67ac6373 gst-libs/gst/audio/gstbaseaudiosink.c: Guard against over and underflows because of clock slaving.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_skew_slaving), (gst_base_audio_sink_render):
Guard against over and underflows because of clock slaving.
When we are using our own clock, still compensate for any calibrations
that we might have done to our clock.
2008-04-03 10:37:03 +00:00
Sebastian Dröge
49467c4a63 gst-libs/gst/cdda/: Use GLib's base64 implementation instead of our own.
Original commit message from CVS:
* gst-libs/gst/cdda/Makefile.am:
* gst-libs/gst/cdda/base64.c:
* gst-libs/gst/cdda/base64.h:
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cddabasesrc_calculate_musicbrainz_discid):
Use GLib's base64 implementation instead of our own.
2008-04-03 06:39:27 +00:00
Wim Taymans
4b7e9b70f7 gst-libs/gst/riff/riff-media.c: ms-gsm can have arbitrarty sample rates. See #481354.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
ms-gsm can have arbitrarty sample rates. See #481354.
2008-03-31 16:08:45 +00:00
Wim Taymans
9ac953ef0f gst-libs/gst/riff/riff-media.c: MP4S is generic MPEG-4, not a microsoft variant.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
MP4S is generic MPEG-4, not a microsoft variant.
2008-03-28 16:22:35 +00:00
Wim Taymans
fc97139103 gst-libs/gst/rtsp/gstrtspconnection.c: Don't error when poll_wait returns EAGAIN.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_connect), (gst_rtsp_connection_write),
(gst_rtsp_connection_read_internal), (gst_rtsp_connection_poll):
Don't error when poll_wait returns EAGAIN.
2008-03-24 16:40:08 +00:00
Wim Taymans
877a45b791 gst-libs/gst/audio/gstaudiosink.c: Small debug improvement.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_release):
Small debug improvement.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
Fix bug in determining the sample start/stop position, we want to base
this decision on the fact that we are going forwards or backwards, not
slower or faster. This fixes some ugly resync warnings when playing at
very slow speeds.
2008-03-24 11:24:22 +00:00
Sebastian Dröge
49deb0c05d Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory u...
Original commit message from CVS:
* configure.ac:
* ext/alsa/gstalsamixerelement.c:
(gst_alsa_mixer_element_class_init):
* ext/alsa/gstalsasink.c: (gst_alsasink_class_init):
* ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init):
* ext/cdparanoia/gstcdparanoiasrc.c:
(gst_cd_paranoia_src_class_init):
* ext/gio/gstgiosink.c: (gst_gio_sink_class_init):
* ext/gio/gstgiosrc.c: (gst_gio_src_class_init):
* ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_class_init):
* ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_class_init):
* ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init):
* ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init):
* ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
* ext/pango/gsttextrender.c: (gst_text_render_class_init):
* ext/theora/theoradec.c: (gst_theora_dec_class_init):
* ext/theora/theoraenc.c: (gst_theora_enc_class_init):
* ext/theora/theoraparse.c: (gst_theora_parse_class_init):
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_class_init):
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
(gst_audio_filter_template_class_init):
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init):
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_class_init):
* gst-libs/gst/interfaces/mixertrack.c:
(gst_mixer_track_class_init):
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_class_init):
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_class_init):
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_class_init):
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_class_init):
* gst/audioresample/gstaudioresample.c:
(gst_audioresample_class_init):
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audio_test_src_class_init):
* gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init):
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
(preroll_unlinked):
* gst/playback/gstplaybin.c: (gst_play_bin_class_init):
* gst/playback/gstplaybin2.c: (gst_play_bin_class_init):
* gst/playback/gstplaysink.c: (gst_play_sink_class_init):
* gst/playback/gstqueue2.c: (gst_queue_class_init):
* gst/playback/gststreaminfo.c: (gst_stream_info_class_init):
* gst/playback/gststreamselector.c: (gst_selector_pad_class_init),
(gst_stream_selector_class_init):
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init):
* gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
* gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
* gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init):
* gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init):
* gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init):
* gst/videorate/gstvideorate.c: (gst_video_rate_class_init):
* gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init):
* gst/videotestsrc/gstvideotestsrc.c:
(gst_video_test_src_class_init):
* gst/volume/gstvolume.c: (gst_volume_class_init):
* sys/v4l/gstv4lelement.c: (gst_v4lelement_class_init):
* sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init):
* sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init):
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init):
* sys/ximage/ximagesink.c: (gst_ximagesink_class_init):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use
static strings (i.e. all). This gives us less memory usage,
fewer allocations and thus less memory defragmentation. Depend
on core CVS for this. Fixes bug #523806.
2008-03-22 15:00:53 +00:00
Wim Taymans
6d1fdf60a3 gst-libs/gst/sdp/gstsdpmessage.c: Use GST_STR_NULL when trying to print strings that could be NULL because this might...
Original commit message from CVS:
* gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_message_dump):
Use GST_STR_NULL when trying to print strings that could be NULL because
this might crash on some platforms. See #520808.
2008-03-18 11:20:05 +00:00
Ole André Vadla Ravnås
aca58fb8ac gst-libs/gst/rtsp/gstrtspconnection.c: Generic Windows fixes that makes libgstrtsp work on Windows when coupled with ...
Original commit message from CVS:
Patch by: Ole André Vadla Ravnås  <ole.andre.ravnas@tandberg.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_connect), (gst_rtsp_connection_write),
(read_line), (gst_rtsp_connection_read_internal):
Generic Windows fixes that makes libgstrtsp work on Windows when
coupled with the new GstPoll API. See #520808.
2008-03-18 11:10:12 +00:00
Tim-Philipp Müller
a0de863603 gst-libs/gst/video/video.c: Fix gst_video_format_parse_caps() for RGB caps with alpha channel (#522635).
Original commit message from CVS:
* gst-libs/gst/video/video.c: (gst_video_format_parse_caps),
(gst_video_format_from_rgba32_masks):
Fix gst_video_format_parse_caps() for RGB caps with alpha channel
(#522635).
* tests/check/libs/video.c: (test_parse_caps_rgb), (video_suite):
Add unit test for the RGB caps parsing and creation, checking for
internal consistency of the new API and consistency of the API with
the old GST_VIDEO_CAPS_* defines.
2008-03-17 10:32:28 +00:00
Michael Smith
15e209d20e gst-libs/gst/audio/gstringbuffer.*: Rename recently added buffer types to make more sense.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps):
* gst-libs/gst/audio/gstringbuffer.h:
Rename recently added buffer types to make more sense.
* ext/alsa/gstalsasink.c: (alsasink_parse_spec),
(gst_alsasink_write):
Adapt for above API changes.
Fixes bug #520523.
2008-03-12 12:39:13 +00:00
Wim Taymans
579949e2c5 gst-libs/gst/audio/gstbaseaudiosrc.c: Fix duration when no clock was provided. Fixes #520300.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
Fix duration when no clock was provided. Fixes #520300.
2008-03-10 17:19:56 +00:00
Olivier Crete
cf273d8add Add trivial function to compare GstNetAddress. See #520626.
Original commit message from CVS:
Patch by: Olivier Crete  <tester at tester ca>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/netbuffer/gstnetbuffer.c: (gst_netaddress_equal):
* gst-libs/gst/netbuffer/gstnetbuffer.h:
Add trivial function to compare GstNetAddress. See #520626.
API: GstNetBuffer::gst_netaddress_equal
2008-03-07 18:17:44 +00:00
Wim Taymans
8a822e70be gst/: Remove GstPollMode from gstpoll constructor.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_create):
* gst/tcp/gstmultifdsink.c: (gst_fdset_mode_get_type),
(gst_multi_fd_sink_class_init), (gst_multi_fd_sink_start):
* gst/tcp/gstmultifdsink.h:
* gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_start):
* gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_start):
Remove GstPollMode from gstpoll constructor.
2008-03-07 15:48:51 +00:00
Mersad Jelacic
ed814cbaed gst-libs/gst/rtp/gstbasertpaudiopayload.*: API: add gst_base_rtp_audio_payload_set_samplebits_options() to make it po...
Original commit message from CVS:
Patch by: Mersad Jelacic  <mersad at axis dot com>
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
API: add gst_base_rtp_audio_payload_set_samplebits_options() to make it
possible to specify the sample size in bits. (#509637)
2008-03-03 16:11:50 +00:00
Tim-Philipp Müller
5a3d087279 API: add gst_mixer_options_list_changed(), gst_mixer_mixer_changed() and gst_mixer_message_parse_options_list_changed...
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/interfaces/mixer.c: (gst_mixer_option_changed),
(gst_mixer_options_list_changed), (gst_mixer_mixer_changed),
(gst_mixer_message_get_type),
(gst_mixer_message_parse_option_changed),
(gst_mixer_message_parse_options_list_changed):
* gst-libs/gst/interfaces/mixer.h: (GstMixerType),
(GST_MIXER_MESSAGE_OPTION_CHANGED),
(GST_MIXER_MESSAGE_OPTIONS_LIST_CHANGED),
(GST_MIXER_MESSAGE_MIXER_CHANGED):
API: add gst_mixer_options_list_changed(), gst_mixer_mixer_changed()
and gst_mixer_message_parse_options_list_changed(). Fixes #519916.
2008-03-03 13:56:38 +00:00
Tim-Philipp Müller
225d9b5d08 gst-libs/gst/interfaces/mixeroptions.*: API: add GstMixerOptions::get_values vfunc (#519906)
Original commit message from CVS:
* gst-libs/gst/interfaces/mixeroptions.c: (gst_mixer_options_init),
(gst_mixer_options_get_values):
* gst-libs/gst/interfaces/mixeroptions.h:
(GST_MIXER_OPTIONS_GET_CLASS), (GstMixerOptionsClass),
(_GstMixerOptions), (_GstMixerOptionsClass):
API: add GstMixerOptions::get_values vfunc (#519906)
2008-03-03 13:50:18 +00:00
Sebastian Dröge
ec7afb6f84 Correct all relevant warnings found by the sparse semantic code analyzer. This include marking several symbols static...
Original commit message from CVS:
* ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_get_type):
* ext/alsa/gstalsasink.c: (set_hwparams):
* ext/alsa/gstalsasrc.c: (set_hwparams):
* ext/gio/gstgio.c: (gst_gio_uri_handler_get_uri):
* ext/ogg/gstoggmux.h:
* ext/ogg/gstogmparse.c:
* gst-libs/gst/audio/audio.c:
* gst-libs/gst/fft/kiss_fft_f64.c: (kiss_fft_f64_alloc):
* gst-libs/gst/pbutils/missing-plugins.c:
(gst_missing_uri_sink_message_new),
(gst_missing_element_message_new),
(gst_missing_decoder_message_new),
(gst_missing_encoder_message_new):
* gst-libs/gst/rtp/gstbasertppayload.c:
* gst-libs/gst/rtp/gstrtcpbuffer.c:
(gst_rtcp_packet_bye_get_reason):
* gst/audioconvert/gstaudioconvert.c:
* gst/audioresample/gstaudioresample.c:
* gst/ffmpegcolorspace/imgconvert.c:
* gst/playback/test.c: (gen_video_element), (gen_audio_element):
* gst/typefind/gsttypefindfunctions.c:
* gst/videoscale/vs_4tap.c:
* gst/videoscale/vs_4tap.h:
* sys/v4l/gstv4lelement.c:
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_get_any_caps):
* sys/v4l/v4l_calls.c:
* sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_capture_init),
(gst_v4lsrc_try_capture):
* sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
(gst_ximagesink_ximage_new):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls),
(gst_xvimagesink_xvimage_new):
* tests/check/elements/audioconvert.c:
* tests/check/elements/audioresample.c:
(fail_unless_perfect_stream):
* tests/check/elements/audiotestsrc.c: (setup_audiotestsrc):
* tests/check/elements/decodebin.c:
* tests/check/elements/gdpdepay.c: (setup_gdpdepay),
(setup_gdpdepay_streamheader):
* tests/check/elements/gdppay.c: (setup_gdppay), (GST_START_TEST),
(setup_gdppay_streamheader):
* tests/check/elements/gnomevfssink.c: (setup_gnomevfssink):
* tests/check/elements/multifdsink.c: (setup_multifdsink):
* tests/check/elements/textoverlay.c:
* tests/check/elements/videorate.c: (setup_videorate):
* tests/check/elements/videotestsrc.c: (setup_videotestsrc):
* tests/check/elements/volume.c: (setup_volume):
* tests/check/elements/vorbisdec.c: (setup_vorbisdec):
* tests/check/elements/vorbistag.c:
* tests/check/generic/clock-selection.c:
* tests/check/generic/states.c: (setup), (teardown):
* tests/check/libs/cddabasesrc.c:
* tests/check/libs/video.c:
* tests/check/pipelines/gio.c:
* tests/check/pipelines/oggmux.c:
* tests/check/pipelines/simple-launch-lines.c:
(simple_launch_lines_suite):
* tests/check/pipelines/streamheader.c:
* tests/check/pipelines/theoraenc.c:
* tests/check/pipelines/vorbisdec.c:
* tests/check/pipelines/vorbisenc.c:
* tests/examples/seek/scrubby.c:
* tests/examples/seek/seek.c: (query_positions_elems),
(query_positions_pads):
* tests/icles/stress-xoverlay.c: (myclock):
Correct all relevant warnings found by the sparse semantic code
analyzer. This include marking several symbols static, using
NULL instead of 0 for pointers and using "foo (void)" instead
of "foo ()" for declarations.
* win32/common/libgstrtp.def:
Add gst_rtp_buffer_set_extension_data to the symbol definition file.
2008-03-03 06:04:31 +00:00
Tim-Philipp Müller
d85688b8a2 gst-libs/gst/rtsp/gstrtspconnection.c: Don't include unix headers unconditionally (fixes #518037).
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspconnection.c:
Don't include unix headers unconditionally (fixes #518037).
2008-03-02 18:32:36 +00:00
Tim-Philipp Müller
843366350e gst-libs/gst/video/video.*: API: add GST_VIDEO_FORMAT_Y41B and GST_VIDEO_FORMAT_Y42B.
Original commit message from CVS:
* gst-libs/gst/video/video.c: (gst_video_calculate_display_ratio),
(gst_video_format_from_fourcc), (gst_video_format_to_fourcc),
(gst_video_format_is_rgb), (gst_video_format_is_yuv),
(gst_video_format_has_alpha), (gst_video_format_get_row_stride),
(gst_video_format_get_pixel_stride),
(gst_video_format_get_component_width),
(gst_video_format_get_component_height),
(gst_video_format_get_component_offset), (gst_video_format_get_size):
* gst-libs/gst/video/video.h: (GST_VIDEO_FORMAT_Y41B),
(GST_VIDEO_FORMAT_Y42B):
API: add GST_VIDEO_FORMAT_Y41B and GST_VIDEO_FORMAT_Y42B.
2008-03-02 18:20:44 +00:00
Tim-Philipp Müller
d3273657f0 gst-libs/gst/video/video.c: YV12 is I420 with swapped components 1 and 2, so the offset of component 1 for I420 shoul...
Original commit message from CVS:
* gst-libs/gst/video/video.c: (gst_video_format_get_component_offset):
YV12 is I420 with swapped components 1 and 2, so the offset of
component 1 for I420 should be the offset for component 2 for YV12
and vice versa.
2008-03-02 18:07:10 +00:00
Julien Moutte
f0154849b0 ext/alsa/gstalsa.c: Probe for IEC958 pcm to detect if we can do SPDIF output.
Original commit message from CVS:
2008-02-29  Julien Moutte  <julien@fluendo.com>

* ext/alsa/gstalsa.c: (gst_alsa_open_iec958_pcm),
(gst_alsa_probe_supported_formats): Probe for IEC958 pcm to
detect
if we can do SPDIF output.
* ext/alsa/gstalsa.h:
* ext/alsa/gstalsasink.c: (set_hwparams), (alsasink_parse_spec),
(gst_alsasink_prepare), (gst_alsasink_close),
(gst_alsasink_write):
* ext/alsa/gstalsasink.h: Initial support for SPDIF.
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_parse_caps):
* gst-libs/gst/audio/gstringbuffer.h: Add non linear buffer
types
to support AC3, EC3 and IEC958 buffers.
2008-02-29 18:44:36 +00:00
Tim-Philipp Müller
ca1b8db04a gst-libs/gst/interfaces/mixer.c: De-cruft and fix message type assertions (NULL is not a really valid mixer message t...
Original commit message from CVS:
* gst-libs/gst/interfaces/mixer.c: (GST_MIXER_MESSAGE_HAS_TYPE),
(gst_mixer_message_parse_mute_toggled),
(gst_mixer_message_parse_record_toggled),
(gst_mixer_message_parse_volume_changed),
(gst_mixer_message_parse_option_changed):
De-cruft and fix message type assertions (NULL is not a really
valid mixer message type string).
2008-02-29 17:59:16 +00:00
David Schleef
3fb3c580b1 gst-libs/gst/riff/riff-media.c: Add Dirac mapping
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: Add Dirac mapping
2008-02-29 00:04:57 +00:00
Wim Taymans
0667334b2d gst-libs/gst/rtsp/gstrtspconnection.*: Use GstPoll for the rtsp connection.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_create), (gst_rtsp_connection_connect),
(gst_rtsp_connection_write), (gst_rtsp_connection_read_internal),
(gst_rtsp_connection_receive), (gst_rtsp_connection_close),
(gst_rtsp_connection_free), (gst_rtsp_connection_poll),
(gst_rtsp_connection_flush):
* gst-libs/gst/rtsp/gstrtspconnection.h:
Use GstPoll for the rtsp connection.
2008-02-28 09:50:52 +00:00
Sebastian Dröge
07cd425c25 gst-libs/gst/cdda/sha1.c: Use memcpy() instead of upcasting a byte array to long *. This fixes an unaligned memory ac...
Original commit message from CVS:
* gst-libs/gst/cdda/sha1.c: (sha_transform):
Use memcpy() instead of upcasting a byte array to long *. This
fixes an unaligned memory access, resulting in SIGBUS on IA64.
This should be ported to GCheckSum once we can use GLib 2.16.
Partially fixes bug #500833.
2008-02-27 10:42:08 +00:00
Tim-Philipp Müller
b191c8f609 gst-libs/gst/tag/gsttagdemux.c: Push tag event after the newsegment event. Log the pointer of the buffer we're actual...
Original commit message from CVS:
* gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_chain):
Push tag event after the newsegment event. Log the pointer of
the buffer we're actually going to push rather than the buffer
we're feeding to _make_metadata_writable().
2008-02-27 10:23:27 +00:00
Tim-Philipp Müller
2c538ea740 gst-libs/gst/audio/multichannel.c: Fix confusing terminology in docs and code: structure fields are 'fields' and not ...
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
(GST_AUDIO_CHANNEL_POSITIONS_FIELD_NAME),
(gst_audio_get_channel_positions), (gst_audio_set_channel_positions),
(gst_audio_set_structure_channel_positions_list),
(add_list_to_struct), (gst_audio_set_caps_channel_positions_list),
(gst_audio_fixate_channel_positions):
Fix confusing terminology in docs and code: structure fields are
'fields' and not 'properties'.
2008-02-19 20:42:09 +00:00
Tim-Philipp Müller
a1e59086ba gst-libs/gst/audio/multichannel.c: Give more useful warning messages if one of the channel layout enums passed to us ...
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions), (add_list_to_struct):
Give more useful warning messages if one of the channel
layout enums passed to us is invalid and if the "channels"
field in the caps has a GType we don't expect.
2008-02-19 20:36:58 +00:00
Tim-Philipp Müller
29162d0a46 gst-libs/gst/audio/multichannel.c: Fix typo in docs blurb.
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
Fix typo in docs blurb.
2008-02-19 20:22:09 +00:00
Wim Taymans
d8c28a99c8 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Fix potential leaks.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_handle_frame_based_buffer),
(gst_base_rtp_audio_payload_handle_sample_based_buffer):
Fix potential leaks.
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_chain):
Fix leak when there is no function configured.
2008-02-12 14:51:26 +00:00
Jan Schmidt
a30f5934a6 gst-libs/gst/interfaces/: Comment out a couple of other things which break the build when
Original commit message from CVS:
* gst-libs/gst/interfaces/mixeroptions.c:
* gst-libs/gst/interfaces/mixertrack.c:
Comment out a couple of other things which break the build when
GST_DISABLE_DEPRECATED isn't on but -Werror is.
2008-02-07 22:39:00 +00:00
Wim Taymans
d4bcdb7340 gst-libs/gst/rtsp/gstrtspmessage.c: Fix compiler warning.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_unset):
Fix compiler warning.
2008-02-07 12:17:49 +00:00
Peter Kjellerstedt
ed373d3072 gst-libs/gst/sdp/gstsdpmessage.c: Clear the addrinfo struct using memset. Fixes #514937.
Original commit message from CVS:
Patch by: Peter Kjellerstedt  <pkj at axis com>
* gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address):
Clear the addrinfo struct using memset. Fixes #514937.
2008-02-07 11:00:45 +00:00
Sebastian Dröge
8b970c5581 Use gmtime_r if available as gmtime is not MT-safe.
Original commit message from CVS:
* configure.ac:
* gst-libs/gst/rtsp/gstrtspconnection.c: (add_date_header):
Use gmtime_r if available as gmtime is not MT-safe.
Fixes bug #511810.
2008-02-02 07:13:15 +00:00
Sebastian Dröge
044f629784 gst-libs/gst/rtsp/gstrtspconnection.c: Cast glong to time_t as time_t might have a different type on other platforms,...
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspconnection.c: (add_date_header):
Cast glong to time_t as time_t might have a different type on
other platforms, like FreeBSD, and we get a compiler warning
otherwise. Fixes bug #511825.
2008-02-02 06:52:41 +00:00
Thijs Vermeir
b8d39bc200 Add gst_rtp_buffer_set_extension_data()
Original commit message from CVS:
Patch by: Thijs Vermeir <thijsvermeir at gmail dot com>
* gst-libs/gst/rtp/gstrtpbuffer.c:
(gst_rtp_buffer_set_extension_data):
* gst-libs/gst/rtp/gstrtpbuffer.h:
* tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
Add gst_rtp_buffer_set_extension_data()
Add a unit test for this addition. Fixes #511478.
API: GstRTPBuffer:gst_rtp_buffer_set_extension_data()
2008-02-01 11:09:16 +00:00
Wim Taymans
88322ce012 gst-libs/gst/app/gstappsink.c: Really clean up the queue instead of just unreffing all buffers in it.
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_dispose):
Really clean up the queue instead of just unreffing all buffers
in it.
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_base_init),
(gst_app_src_class_init), (gst_app_src_init),
(gst_app_src_dispose), (gst_app_src_finalize):
Fix dispose/finalize.
2008-01-31 17:18:46 +00:00
Jan Schmidt
67ab0ec286 gst-libs/gst/interfaces/: Also remove the conditional registration of the signals that disappeared with the ABI chang...
Original commit message from CVS:
* gst-libs/gst/interfaces/mixeroptions.c:
* gst-libs/gst/interfaces/mixertrack.c:
Also remove the conditional registration of the signals
that disappeared with the ABI change in 0.10.14
2008-01-30 13:45:27 +00:00
Jan Schmidt
2531fca577 gst-libs/gst/rtsp/gstrtspconnection.c: Revert patch to gstrtspconnection.c for brown paper bag release of -base. Re-o...
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspconnection.c:
Revert patch to gstrtspconnection.c for brown paper bag
release of -base. Re-opens: #511825
2008-01-30 12:28:59 +00:00
Jan Schmidt
bb132eda11 gst-libs/gst/interfaces/: Change the way these deprecated function pointers are removed so that the compiled ABI is u...
Original commit message from CVS:
* gst-libs/gst/interfaces/mixeroptions.h:
* gst-libs/gst/interfaces/mixertrack.h:
Change the way these deprecated function pointers are removed
so that the compiled ABI is unconditionally smaller. This
sets in stone an ABI break that actually occurred when the
things were deprecated in 0.10.14, which seems to be the best
fix as the only known users are oss-mixer and sunaudio-mixer in
gst-plugins-good.
2008-01-30 12:19:02 +00:00
Sebastian Dröge
bd4bb423e9 gst-libs/gst/rtsp/gstrtspconnection.c: Cast glong to time_t as time_t might have a different type on other platforms,...
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspconnection.c: (add_date_header):
Cast glong to time_t as time_t might have a different type on
other platforms, like FreeBSD, and we get a compiler warning
otherwise. Fixes bug #511825.
2008-01-29 09:59:03 +00:00
Sebastian Dröge
a6e4222c70 gst-libs/gst/audio/gstaudiofilter.c: Initialize the GstRingerBuffer class to get it's debug category initialized. gst...
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiofilter.c:
(gst_audio_filter_class_init):
Initialize the GstRingerBuffer class to get it's debug category
initialized. gst_ring_buffer_parse_caps() uses the ringbuffer debug
category and otherwise we get some g_critical(). Fixes bug #512334.
2008-01-29 09:47:12 +00:00
Thijs Vermeir
d821d8b783 gst-libs/gst/rtp/gstrtpbuffer.c: Fix typos and wrong extension check. Fixes #511274.
Original commit message from CVS:
Patch by: Thijs Vermeir <thijsvermeir at gmail dot com>
* gst-libs/gst/rtp/gstrtpbuffer.c:
(gst_rtp_buffer_get_extension_data):
Fix typos and wrong extension check. Fixes #511274.
2008-01-22 15:37:49 +00:00
Sébastien Moutte
db4d879623 gst-libs/gst/sdp/gstsdpmessage.c: Include Winsock2.h for VS6 and use a different way initialize hints structure so it...
Original commit message from CVS:
* gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address):
Include Winsock2.h for VS6 and use a different way initialize
hints structure so it can build with VS6.
* win32/MANIFEST:
* win32/vs6/libgstsdp.dsp:
* win32/common/libgstsdp.def:
Add new files for libgstsdp.
* win32/vs6/grammar.dsp:
Copy pbutils-enumtypes* from win32/common to pbutils sources folder.
* win32/vs6/gst_plugins_base.dsw:
* win32/vs6/libgstdecodebin.dsp:
* win32/vs6/libgstdecodebin2.dsp:
* win32/vs6/libgstplaybin.dsp:
* win32/vs6/libgstvolume.dsp:
Add new dependencies to the link list.
2008-01-13 21:40:45 +00:00
Jan Schmidt
fa2f3fb98e Add test to see if hstrerror is available or if we need libresolv (Solaris) for it, then use it in libgstrtsp.
Original commit message from CVS:
* configure.ac:
* gst-libs/gst/rtsp/Makefile.am:
Add test to see if hstrerror is available or if we need libresolv
(Solaris) for it, then use it in libgstrtsp.
2008-01-12 23:08:28 +00:00
Jan Schmidt
c571cc1e63 gst-libs/gst/tag/Makefile.am: Fix include path order
Original commit message from CVS:
* gst-libs/gst/tag/Makefile.am:
Fix include path order
2008-01-12 14:54:51 +00:00
Tim-Philipp Müller
145bd67812 Ignore more and make buildbot happy
Original commit message from CVS:
Ignore more and make buildbot happy
2008-01-11 17:15:23 +00:00
Edward Hervey
8efa9193a3 gst-libs/gst/pbutils/install-plugins.*: Add GBoxed GType for GstInstallPluginsContext, this eases the wrapping for bi...
Original commit message from CVS:
* gst-libs/gst/pbutils/install-plugins.c:
(gst_install_plugins_context_copy),
(gst_install_plugins_context_get_type):
* gst-libs/gst/pbutils/install-plugins.h:
Add GBoxed GType for GstInstallPluginsContext, this eases the wrapping
for bindings.
2008-01-11 16:18:10 +00:00
Tim-Philipp Müller
3feb4bc8c5 gst-libs/gst/audio/: Ref audio clock class from a thread-safe context to make sure however unlikely that may be in pr...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init):
Ref audio clock class from a thread-safe context to make sure
we're not bit by GObjects lack of thread-safety here (#349410),
however unlikely that may be in practice.
2008-01-10 17:55:53 +00:00
Tim-Philipp Müller
526fd3d7da gst-libs/gst/tag/gsttagdemux.c: Post an error message if we can't pull as many bytes as we need for the tag. This mak...
Original commit message from CVS:
* gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_pull_start_tag):
Post an error message if we can't pull as many bytes as we need
for the tag. This makes sure the user gets to see a proper error
message if a file with a partial ID3 tag is fed to decodebin, and
not a 'no ID3 tag demuxer' error, which would be confusing
(see #508138).
2008-01-08 21:10:02 +00:00
Tim-Philipp Müller
939d39cdf4 gst-libs/gst/pbutils/descriptions.c: Add description strings for ID3, APE, and ICY tags.
Original commit message from CVS:
* gst-libs/gst/pbutils/descriptions.c: (formats):
Add description strings for ID3, APE, and ICY tags.
2008-01-08 20:59:20 +00:00
Sebastian Dröge
a000758477 gst-libs/gst/audio/gstaudiofilter.c: Don't set element details for the abstract GstAudioFilter class.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_get_type):
Don't set element details for the abstract GstAudioFilter class.
2008-01-03 07:17:05 +00:00
Sebastian Dröge
0e5857ea26 gst-libs/gst/audio/gstaudiofilter.c: Implement get_unit_size() vmethod of GstBaseTransform.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiofilter.c:
(gst_audio_filter_class_init), (gst_audio_filter_get_unit_size):
Implement get_unit_size() vmethod of GstBaseTransform.
2008-01-02 12:09:48 +00:00
Edward Hervey
7701190c01 gst-libs/gst/pbutils/: Use glib-enum generator to have a proper enum GType for
Original commit message from CVS:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/pbutils/pbutils.h:
Use glib-enum generator to have a proper enum GType for
GST_TYPE_INSTALL_PLUGINS_RETURN so we can easily wrap it in bindings.
2008-01-01 12:53:48 +00:00
Wim Taymans
db8c746760 gst-libs/gst/rtsp/gstrtspdefs.*: Add Location header so that we can start implementing redirects.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status):
* gst-libs/gst/rtsp/gstrtspdefs.h:
Add Location header so that we can start implementing redirects.
See #506025.
2007-12-30 19:21:16 +00:00
David Schleef
d605ca3ba2 gst-libs/gst/video/video.*: Add additional RGBA and RGB-24 video formats.
Original commit message from CVS:
* gst-libs/gst/video/video.c:
* gst-libs/gst/video/video.h:
Add additional RGBA and RGB-24 video formats.
2007-12-22 05:19:00 +00:00
Thijs Vermeir
169259294f gst-libs/gst/riff/riff-read.c: Use GST_ROUND_UP_2 macro
Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c:
Use GST_ROUND_UP_2 macro
2007-12-20 19:43:25 +00:00
Wim Taymans
355e8a940d gst-libs/gst/audio/gstaudiosink.c: Improve debug output.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
(gst_audioringbuffer_open_device),
(gst_audioringbuffer_close_device), (gst_audioringbuffer_acquire),
(gst_audioringbuffer_release), (gst_audioringbuffer_start),
(gst_audioringbuffer_pause), (gst_audioringbuffer_stop),
(gst_audio_sink_create_ringbuffer):
Improve debug output.
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_start),
(gst_ring_buffer_pause), (gst_ring_buffer_delay):
Prevent some functions from doing things and failing when the
ringbuffer is not yet acquired.
2007-12-18 15:56:51 +00:00
Sebastian Dröge
c25cb3fc98 gst-libs/gst/interfaces/interfaces.h: Also remove interfaces.h from CVS as it is not needed anymore.
Original commit message from CVS:
* gst-libs/gst/interfaces/interfaces.h:
Also remove interfaces.h from CVS as it is not needed anymore.
2007-12-18 15:32:49 +00:00
Sebastian Dröge
ee3edd8901 gst-libs/gst/interfaces/Makefile.am: interfaces.h is not used anymore so remove it from the build process.
Original commit message from CVS:
* gst-libs/gst/interfaces/Makefile.am:
interfaces.h is not used anymore so remove it from the build
process.
2007-12-18 15:20:12 +00:00
David Schleef
1c3e012fc3 Add new GstVideFormat enum and write a bunch of helper functions based around it.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/video/video.c:
* gst-libs/gst/video/video.h:
Add new GstVideFormat enum and write a bunch of helper functions
based around it.
2007-12-18 00:13:26 +00:00
Wim Taymans
2ea251a366 gst-libs/gst/audio/gstbaseaudiosrc.c: Add debug info.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create), (gst_base_audio_src_change_state):
Add debug info.
When going from PLAYING to PAUSED, pause the ringbuffer before calling
the parent state change function, just like the audiosink, because the
parent waits for the element to finish its processing before completing
the state change. This makes going to PAUSED a lot snappier.
When going from READY to PAUSED, don't allow the ringbuffer to start
yet.
2007-12-17 16:44:51 +00:00
Edward Hervey
80d1fbac43 gst-libs/gst/riff/riff-media.c: Yet another fix for broken software that produce files with an empty blockalign field...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Yet another fix for broken software that produce files with an empty
blockalign field. Instead of completely failing, make a second attempt
at guessing the width/depth by looking at strf->size.
2007-12-17 00:01:00 +00:00
Tim-Philipp Müller
aa47d077c1 gst-libs/gst/: Turn a few g_assert_not_reached() into g_return_val_if_reached() to avoid compiler warnings (#503930).
Original commit message from CVS:
* gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_do_seek),
(gst_cdda_base_src_handle_track_seek), (gst_cdda_base_src_create):
* gst-libs/gst/pbutils/install-plugins.c:
(gst_install_plugins_spawn_child), (gst_install_plugins_supported):
* gst-libs/gst/pbutils/missing-plugins.c:
(gst_missing_plugin_message_get_installer_detail),
(gst_missing_encoder_installer_detail_new):
* gst-libs/gst/rtsp/gstrtspconnection.c: (gst_rtsp_connection_send):
* gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_unset):
Turn a few g_assert_not_reached() into g_return_val_if_reached() to
avoid compiler warnings (#503930).
2007-12-16 23:52:58 +00:00
Edward Hervey
ea931c47d6 gst-libs/gst/riff/riff-media.c: Be apologetic of software that use the 'jpeg' instead of 'JPEG' FOURCC for jpeg video...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
Be apologetic of software that use the 'jpeg' instead of 'JPEG' FOURCC
for jpeg video streams.
Add the 'avc1'/'AVC1' fourcc mapping for h264, same software-comment as
for the above modification.
2007-12-16 23:46:16 +00:00
Tim-Philipp Müller
d23f2ae402 gst-libs/gst/interfaces/xoverlay.c: More guards (we don't want klass to end up being NULL).
Original commit message from CVS:
* gst-libs/gst/interfaces/xoverlay.c: (gst_x_overlay_expose),
(gst_x_overlay_handle_events):
More guards (we don't want klass to end up being NULL).
2007-12-15 17:27:48 +00:00
Peter Kjellerstedt
ba4bb0fa8c gst-libs/gst/rtsp/gstrtspconnection.c: Close control sockets. Fixes #503440.
Original commit message from CVS:
Patch by: Peter Kjellerstedt  <pkj at axis com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_free):
Close control sockets. Fixes #503440.
2007-12-13 15:54:00 +00:00
Tim-Philipp Müller
41e54ce132 gst-libs/gst/pbutils/descriptions.c: Add description for 'private' dts caps (who come up with that name?).
Original commit message from CVS:
* gst-libs/gst/pbutils/descriptions.c: (formats):
Add description for 'private' dts caps (who come up with that name?).
2007-12-13 11:40:10 +00:00
Tim-Philipp Müller
b9cbedfb59 Makefile.am: Add check-exports target and run it with 'make check'.
Original commit message from CVS:
* Makefile.am:
Add check-exports target and run it with 'make check'.
* configure.ac:
Be stricter about what we export in our libraries: change regexp so that
we only export _gst_foo(), but not __gst_foo().
* gst-libs/gst/cdda/base64.h: (rfc822_binary):
* gst-libs/gst/cdda/sha1.h: (sha_init), (sha_update), (sha_final):
Change internal functions to __gst_foo so they dont' get exported.
* win32/common/libgstaudio.def:
Add missing symbols.
2007-12-13 10:10:35 +00:00
Wim Taymans
74170deee7 gst-libs/gst/riff/riff-media.c: No need for floating point operations here. avoids having to link against the math li...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
No need for floating point operations here. avoids having to link
against the math library too.
2007-12-10 15:13:55 +00:00
Tim-Philipp Müller
6623ddbfbd Add one or two missing formats. Generate ADPCM description dynamically depending on layout/format.
Original commit message from CVS:
* gst-libs/gst/pbutils/descriptions.c: (formats),
(format_info_get_desc):
* tests/check/libs/pbutils.c: (GST_START_TEST), (caps_strings),
(GST_START_TEST):
Add one or two missing formats.  Generate ADPCM description
dynamically depending on layout/format.
2007-12-10 11:16:25 +00:00
Wim Taymans
75aa042450 gst-libs/gst/rtp/gstbasertppayload.c: Use runnning time as the base time instead of the timestamp.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
Use runnning time as the base time instead of the timestamp.
Spotted by Saur on IRC.
2007-12-03 13:47:00 +00:00
Edward Hervey
ab433ee7c9 gst-libs/gst/riff/riff-media.c: Add 'WVC1' codec mapping for Windows Media VC-1 video codec.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
Add 'WVC1' codec mapping for Windows Media VC-1 video codec.
2007-12-03 11:32:30 +00:00
Wim Taymans
ac1cc82165 gst-libs/gst/audio/gstbaseaudiosink.c: Our EOS time contains the base_time, _wait_eos() expects a running_time so we ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain):
Our EOS time contains the base_time, _wait_eos() expects a running_time
so we have to subtract the base_time again before calling the function.
This fixes an EOS regression where the base_time was added twice and EOS
took longer and longer in certain situations.
Fixes #498767.
2007-11-21 18:02:21 +00:00
Wim Taymans
157a65b15e Expose methods for some object properties so that subclasses can more easily configure them.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/gstbaseaudiosink.c: (slave_method_get_type),
(gst_base_audio_sink_set_provide_clock),
(gst_base_audio_sink_get_provide_clock),
(gst_base_audio_sink_set_slave_method),
(gst_base_audio_sink_get_slave_method),
(gst_base_audio_sink_set_property),
(gst_base_audio_sink_get_property), (gst_base_audio_sink_drain),
(gst_base_audio_sink_none_slaving),
(gst_base_audio_sink_handle_slaving):
* gst-libs/gst/audio/gstbaseaudiosink.h:
Expose methods for some object properties so that subclasses can more
easily configure them.
Added slave method none, that completely disables slaving to the
internal clock.
API: gst_base_audio_sink_set_provide_clock()
API: gst_base_audio_sink_get_provide_clock()
API: gst_base_audio_sink_set_slave_method()
API: gst_base_audio_sink_get_slave_method()
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_set_provide_clock),
(gst_base_audio_src_get_provide_clock),
(gst_base_audio_src_set_property),
(gst_base_audio_src_get_property), (gst_base_audio_src_create):
* gst-libs/gst/audio/gstbaseaudiosrc.h:
Expose methods for some object properties so that subclasses can more
easily configure them.
API: gst_base_audio_src_set_provide_clock()
API: gst_base_audio_src_get_provide_clock()
2007-11-21 13:04:17 +00:00
Joe Peterson
bd8885220e gst-libs/gst/sdp/gstsdpmessage.c: Fix compilation on FreeBSD (Gentoo). Fixes #498228.
Original commit message from CVS:
Patch by: Joe Peterson <lavajoe at gentoo dot org>
* gst-libs/gst/sdp/gstsdpmessage.c:
Fix compilation on FreeBSD (Gentoo). Fixes #498228.
2007-11-20 07:53:56 +00:00
Tommi Myöhänen
385ed6384f gst-libs/gst/rtsp/gstrtspmessage.c: Fix leaking headers. Fixes #496761.
Original commit message from CVS:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_unset):
Fix leaking headers. Fixes #496761.
2007-11-16 11:22:09 +00:00
Tim-Philipp Müller
dffc29eb13 gst-libs/gst/tag/gstid3tag.c: Add mapping for audio cd discid tags, so we can extract them from tags as well (see #34...
Original commit message from CVS:
* gst-libs/gst/tag/gstid3tag.c: (user_tag_matches),
(gst_tag_from_id3_user_tag):
Add mapping for audio cd discid tags, so we can extract
them from tags as well (see #347848). Also compare identifiers
in ID3v2 TXXX frames in a case-insensitive way to increase
compatibility when reading tags (discid vs. DiscID vs. DiscId).
2007-11-16 10:14:34 +00:00
Jan Schmidt
31f97ca497 gst-libs/gst/fft/: Don't include malloc.h which doesn't exist on Mac OSX.
Original commit message from CVS:
* gst-libs/gst/fft/kiss_fft_f32.h:
* gst-libs/gst/fft/kiss_fft_f64.h:
* gst-libs/gst/fft/kiss_fft_s16.h:
* gst-libs/gst/fft/kiss_fft_s32.h:
Don't include malloc.h which doesn't exist on Mac OSX.
Instead, pull in glib.h and use g_malloc/g_free for
consistency. Fixes: #496548
2007-11-14 11:08:48 +00:00
Sebastien Moutte
13d89656c0 gst-libs/gst/rtp/: Fix some C99-isms and and a missing function that some versions of
Original commit message from CVS:
Patch by: Sebastien Moutte  <sebastien moutte net>
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_ntp_to_unix),
(gst_rtcp_unix_to_ntp):
* gst-libs/gst/rtp/gstrtppayloads.c: (gst_rtp_payload_info_for_name):
Fix some C99-isms and and a missing function that some versions of
MSVC don't like too much (#494346).
* win32/vs6/gst_plugins_base.dsw:
* win32/vs6/libgstaudio.dsp:
* win32/vs6/libgstrtp.dsp:
* win32/vs6/libgsttag.dsp:
Update vs6 projects files (#494346).
2007-11-06 23:35:39 +00:00
Sebastian Dröge
edb4a505d7 Remove the magnitude and phase calculation functions as these have very special use cases and can't even be used for ...
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/fft/gstfftf32.c:
* gst-libs/gst/fft/gstfftf32.h:
* gst-libs/gst/fft/gstfftf64.c:
* gst-libs/gst/fft/gstfftf64.h:
* gst-libs/gst/fft/gstffts16.c:
* gst-libs/gst/fft/gstffts16.h:
* gst-libs/gst/fft/gstffts32.c:
* gst-libs/gst/fft/gstffts32.h:
* tests/check/libs/fft.c: (GST_START_TEST):
Remove the magnitude and phase calculation functions as these have
very special use cases and can't even be used for the spectrum
element. Also adjust the docs to mention some properties of the used
FFT implemention, i.e. how the values are scaled. Fixes #492098.
2007-11-06 11:58:59 +00:00
Vincent Torri
be04e7c294 gst-libs/gst/fft/: Include our own _stdint.h instead of sys/types.h, makes MingW happy (#492306).
Original commit message from CVS:
Patch by: Vincent Torri <vtorri at univ-evry dot fr>
* gst-libs/gst/fft/kiss_fft_s16.h: (KISS_FFT_S16_MALLOC):
* gst-libs/gst/fft/kiss_fft_s32.h: (KISS_FFT_S32_MALLOC):
Include our own _stdint.h instead of sys/types.h, makes MingW happy
(#492306).
* gst-libs/gst/rtsp/gstrtspconnection.c: (gst_rtsp_connection_create):
Use _pipe directly, GLib doesn't have a pipe() macro any longer
(it disappeared in GLib 2.14.0) (#492306).
* gst-libs/gst/sdp/Makefile.am:
* gst-libs/gst/sdp/gstsdpmessage.c:
Fix includes and LIBS for win32/Mingw (#492306).
* tests/examples/dynamic/addstream.c (pause_play_stream):
Use more portable g_usleep() instead of sleep() (#492306).
2007-11-01 13:28:59 +00:00
Ole André Vadla Ravnås
05a205860d gst-libs/gst/audio/gstringbuffer.c: Return NULL instead of an enum that happens to be 0, fixes warning on MSVC (#4921...
Original commit message from CVS:
Patch by: Ole André Vadla Ravnås  <ole.andre.ravnas@tandberg.com>
* gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
(gst_ring_buffer_parse_caps):
Return NULL instead of an enum that happens to be 0, fixes warning
on MSVC (#492114).
* gst-libs/gst/audio/gstringbuffer.h:
No trailing commas in enum list (for gcc-2.9x).
* gst/videotestsrc/videotestsrc.c: (random_char):
Make information loss explicit instead of implicitly truncating to
eight bits via the return value.  Fixes runtime error on MSVC when
using the debug CRT (#492114).
* win32/common/config.h.in:
Fix a bunch of '#undef FOO bar', which MSVC doesn't like (#492114).
* win32/common/libgstinterfaces.def:
* win32/common/libgstrtp.def:
Export a few more symbols (#492114).
2007-11-01 12:51:57 +00:00
Stefan Kost
28b46c1e5d gst-libs/gst/audio/audio.*: Readd the deprecation guards, but preserve compilability.
Original commit message from CVS:
* gst-libs/gst/audio/audio.c:
* gst-libs/gst/audio/audio.h:
Readd the deprecation guards, but preserve compilability.
2007-11-01 08:06:13 +00:00
Tim-Philipp Müller
55a3eaafea gst-libs/gst/audio/audio.h: Don't guard deprecated enum with #ifndef DISABLE_DEPRECATED, or compilation with DISABLE_...
Original commit message from CVS:
* gst-libs/gst/audio/audio.h:
Don't guard deprecated enum with #ifndef DISABLE_DEPRECATED, or
compilation with DISABLE_DEPRECATED and without REMOVE_DEPRECATED
(ie. normal cvs builds) will fail.
2007-10-31 15:30:15 +00:00
Stefan Kost
e37568c196 tell gtk-doc about the deprecation guard. Apply more doc fixes.
Original commit message from CVS:
* docs/libs/Makefile.am:
* gst-libs/gst/audio/audio.c:
* gst-libs/gst/audio/audio.h:
* gst-libs/gst/interfaces/mixer.c:
tell gtk-doc about the deprecation guard. Apply more doc fixes.
2007-10-31 12:47:41 +00:00
Stefan Kost
ffa52e2eac Fix the docs according to what gtk-doc complained about.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/gstaudiofilter.h:
* gst-libs/gst/interfaces/mixer.h:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
* gst-libs/gst/rtp/gstbasertpdepayload.h:
* gst-libs/gst/sdp/gstsdpmessage.c:
Fix the docs according to what gtk-doc complained about.
2007-10-30 20:32:14 +00:00
Sebastian Dröge
a81de45df3 gst-libs/gst/riff/riff-media.c: Use the ALSA channel layout as default for wav files without channel layout informati...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_wavext_add_channel_layout),
(gst_riff_wave_add_default_channel_layout),
(gst_riff_wavext_get_default_channel_mask),
(gst_riff_create_audio_caps):
Use the ALSA channel layout as default for wav files without channel
layout information. This fixes playback of chan-id.wav on 5.1 systems
for example. Also refactor the channel layout setting a bit and add
more default channel orders. Fixes #489010.
2007-10-28 11:53:36 +00:00
Tim-Philipp Müller
698e4050b3 gst-libs/gst/tag/: Deprecate GST_TAG_MUSICBRAINZ_SORTNAME, replaced by the newly-added
Original commit message from CVS:
* gst-libs/gst/tag/tag.h: (GST_TAG_MUSICBRAINZ_SORTNAME):
* gst-libs/gst/tag/tags.c:
Deprecate GST_TAG_MUSICBRAINZ_SORTNAME, replaced by the newly-added
GST_TAG_ARTIST_SORTNAME (in an API and ABI compatible way).
* gst-libs/gst/tag/gstid3tag.c: (tag_matches):
Map ID3v2 TSOP, TSOA and TSOT frames to new SORTNAME tags (#414539).
* gst-libs/gst/tag/gstvorbistag.c: (tag_matches),
(gst_tag_to_vorbis_comments):
Map new SORTNAME tags (these tags aren't even semi-official, so I'm
just mapping everything I found in the wild) (#414539).
2007-10-25 15:10:59 +00:00
Tim-Philipp Müller
d33e6ca818 gst-libs/gst/tag/gsttagdemux.c: Don't abort with an assertion if we receive a seek event with a start type of NONE (s...
Original commit message from CVS:
* gst-libs/gst/tag/gsttagdemux.c:
Don't abort with an assertion if we receive a seek event with
a start type of NONE (see launchpad bug #155878).
2007-10-23 14:23:14 +00:00
Wim Taymans
6a20747e83 gst-libs/gst/audio/gstbaseaudiosink.c: Also explicitly release the ringbuffer when going to NULL because it is requir...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain), (gst_base_audio_sink_change_state):
Also explicitly release the ringbuffer when going to NULL because it
is required in the setcaps function, before the state change to PAUSED
completes.
2007-10-16 15:33:31 +00:00
Tim-Philipp Müller
689ca135f3 Extract vorbis comment LICENSE tags correctly.
Original commit message from CVS:
* gst-libs/gst/tag/gstvorbistag.c:
* tests/check/libs/tag.c:
Extract vorbis comment LICENSE tags correctly.
2007-10-11 18:24:09 +00:00
Jason Kivlighn
054e43aa3c Map ID3v2 WCOP frame to GST_TAG_COPYRIGHT_URI (#447000).
Original commit message from CVS:
Patch by: Jason Kivlighn  <jkivlighn gmail com>
* gst-libs/gst/tag/gstid3tag.c:
* tests/check/libs/tag.c:
Map ID3v2 WCOP frame to GST_TAG_COPYRIGHT_URI (#447000).
2007-10-11 16:12:21 +00:00
Tim-Philipp Müller
8458a74526 gst-libs/gst/tag/gsttagdemux.c: Don't error out when a buggy downstream element doesn't handle the newsegment event w...
Original commit message from CVS:
* gst-libs/gst/tag/gsttagdemux.c:
Don't error out when a buggy downstream element doesn't
handle the newsegment event we send properly (especially
not without posting a meaningful error message on the
bus). See bug #471370 and launchpad bug #136264.
2007-10-10 17:01:51 +00:00
Wim Taymans
02f280a9a0 gst-libs/gst/audio/gstbaseaudiosink.c: Use new basesink method to make our EOS drain interruptable.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain):
Use new basesink method to make our EOS drain interruptable.
2007-10-10 15:36:56 +00:00
Jan Schmidt
c96877779a gst-libs/gst/rtp/gstrtppayloads.c: Fix silly search-replace oversight.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtppayloads.c:
Fix silly search-replace oversight.
2007-10-10 09:37:09 +00:00
Laurent Glayal
0f3c871859 gst-libs/gst/rtp/gstbasertppayload.c: Fix caps memleak. Fixes #484989.
Original commit message from CVS:
Patch by: Laurent Glayal <spglegle at yahoo dot fr>
* gst-libs/gst/rtp/gstbasertppayload.c: (copy_fixed),
(gst_basertppayload_set_outcaps):
Fix caps memleak. Fixes #484989.
2007-10-09 09:57:17 +00:00
Wim Taymans
4f707e2172 gst-libs/gst/rtp/gstbasertpdepayload.c: Fix debug output.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain):
Fix debug output.
2007-10-08 18:04:34 +00:00
Wim Taymans
c3dda05a8b gst-libs/gst/audio/gstbaseaudiosrc.c: Also handle the case where there is no clock set on the audio source, like in t...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
Also handle the case where there is no clock set on the audio source,
like in the unit tests.
2007-10-08 18:02:53 +00:00
Jan Schmidt
41ae026041 gst-libs/gst/rtp/gstrtppayloads.c: Use unsigned G_MAXUINT8 instead of -1 to initialise a guint8 to avoid compiler war...
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtppayloads.c:
Use unsigned G_MAXUINT8 instead of -1 to initialise a guint8
to avoid compiler warnings
2007-10-08 17:40:17 +00:00
Tim-Philipp Müller
d3648ca534 gst-libs/gst/tag/gsttagdemux.c: Don't leak caps.
Original commit message from CVS:
* gst-libs/gst/tag/gsttagdemux.c:
Don't leak caps.
2007-10-06 16:49:55 +00:00
Tim-Philipp Müller
f277cd8801 gst-libs/gst/tag/: API: add GstTagDemux base class for simple tag demuxers.
Original commit message from CVS:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/tag/gsttagdemux.c:
* gst-libs/gst/tag/gsttagdemux.h:
API: add GstTagDemux base class for simple tag demuxers.
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
Add GstTagDemux to docs.
2007-10-06 15:04:53 +00:00
Sebastian Dröge
6e51063659 gst-libs/gst/rtp/gstrtpbuffer.c: Fix bug introduced with last commit which inverted the logic and caused all buffers ...
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtpbuffer.c:
(gst_rtp_buffer_get_payload_subbuffer):
Fix bug introduced with last commit which inverted the logic and
caused all buffers to be dropped. Fixes #483620.
Thanks to Laurent Glayal <spglegle at yahoo dot fr> for noticing.
2007-10-05 07:49:25 +00:00
Stefan Kost
ff01787e4e gst-libs/gst/rtp/gstrtpbuffer.c: with regular return and warning.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtpbuffer.c:
Replace g_return_if_val (as it could be disabled), with regular return
and warning.
2007-10-04 06:50:53 +00:00
Wim Taymans
5ba1ed3a21 gst-libs/gst/audio/gstbaseaudiosink.c: When slaved to the clock, don't try to align a sample with the previous one wh...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_async_play):
When slaved to the clock, don't try to align a sample with the previous
one when going to PLAYING again.
2007-10-02 11:11:13 +00:00
Wim Taymans
767366a1e1 gst-libs/gst/sdp/gstsdpmessage.h: Add RFC 3556 bandwidth modifiers.
Original commit message from CVS:
* gst-libs/gst/sdp/gstsdpmessage.h:
Add RFC 3556 bandwidth modifiers.
2007-10-01 16:33:00 +00:00
Wim Taymans
d874f65f6e Update documentation.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtp/gstrtppayloads.c:
Update documentation.
2007-10-01 13:37:31 +00:00
Wim Taymans
7cdfb6d154 gst-libs/gst/rtp/: Added new file and header to deal with payload info.
Original commit message from CVS:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/rtp/gstrtppayloads.c: (gst_rtp_payload_info_for_pt),
(gst_rtp_payload_info_for_name):
* gst-libs/gst/rtp/gstrtppayloads.h:
Added new file and header to deal with payload info.
* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data),
(gst_rtp_buffer_default_clock_rate):
* gst-libs/gst/rtp/gstrtpbuffer.h:
Payload specific stuff is move to new headers.
Implement _default_clock rate using the new payload function.
* gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address),
(gst_sdp_parse_line):
* gst-libs/gst/sdp/gstsdpmessage.h:
Add some more comments.
2007-10-01 13:22:14 +00:00
Tim-Philipp Müller
3263d790f9 Add mappings for the new GST_TAG_COMPOSER for vorbis comments and ID3v2 tags.
Original commit message from CVS:
* configure.ac:
* gst-libs/gst/tag/gstid3tag.c:
* gst-libs/gst/tag/gstvorbistag.c:
Add mappings for the new GST_TAG_COMPOSER for vorbis comments
and ID3v2 tags.
2007-09-25 19:06:47 +00:00
Tim-Philipp Müller
3f7d77fc29 gst-libs/gst/floatcast/floatcast.h: Don't include config.h in an installed public header, this might break compilatio...
Original commit message from CVS:
* gst-libs/gst/floatcast/floatcast.h:
Don't include config.h in an installed public header, this
might break compilation of applications that don't have such
a header and doesn't necessarily do what it's supposed to do
anyway (ie. check for the lrint/lrintf defines) (#442065).
Add docs for the various macros and document how this header
has to be used (link against libm, etc.); add a few FIXMEs;
include math.h for non-c99 code path.  Based on patch by
Jan Schmidt.
2007-09-25 11:54:09 +00:00
Wim Taymans
0bba976bd3 gst-libs/gst/app/gstappsink.*: Add properties, signals and actions to access the element even without linking to the ...
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (gst_app_marshal_OBJECT__VOID),
(gst_app_sink_class_init), (gst_app_sink_init),
(gst_app_sink_dispose), (gst_app_sink_finalize),
(gst_app_sink_set_property), (gst_app_sink_get_property),
(gst_app_sink_flush_unlocked), (gst_app_sink_start),
(gst_app_sink_event), (gst_app_sink_getcaps),
(gst_app_sink_set_caps), (gst_app_sink_get_caps),
(gst_app_sink_is_eos), (gst_app_sink_pull_preroll),
(gst_app_sink_pull_buffer):
* gst-libs/gst/app/gstappsink.h:
Add properties, signals and actions to access the element even without
linking to the library.
Fix some method names and signatures.
2007-09-20 13:59:50 +00:00
Wim Taymans
2f0a269ddd gst-libs/gst/rtp/gstbasertpdepayload.c: Only copy timestamp on outgoing packets if the depayloader did not set one.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_chain),
(gst_base_rtp_depayload_set_gst_timestamp):
Only copy timestamp on outgoing packets if the depayloader did not set
one.
Also copy duration on outgoing packets.
2007-09-19 16:09:57 +00:00
Wim Taymans
70ad18deee gst-libs/gst/rtp/gstbasertppayload.c: Fix compilation because of missing %d in printf.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c: (copy_fixed),
(gst_basertppayload_set_outcaps):
Fix compilation because of missing %d in printf.
When fixating caps, fixate what we can and throw away all remaining
unfixed caps, subclasses should do something smart if they need to.
2007-09-19 15:55:08 +00:00
Jan Schmidt
d5996e9c37 Fix a bunch of compile warnings shown with Forte.
Original commit message from CVS:
* ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
(gst_text_overlay_set_property):
* ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_ntp_to_unix),
(gst_rtcp_unix_to_ntp):
* gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_get_type):
* gst/playback/gstqueue2.c:
* tests/examples/seek/seek.c: (set_scale):
Fix a bunch of compile warnings shown with Forte.
* gst/audiorate/gstaudiorate.c:
Always pull in config.h before including any system headers.
2007-09-17 17:24:55 +00:00
Wim Taymans
523fd097e6 gst-libs/gst/rtp/gstbasertpdepayload.c: Remove code to deal with RTP to GST time conversion, we now just copy the GST...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_init), (gst_base_rtp_depayload_setcaps),
(gst_base_rtp_depayload_chain),
(gst_base_rtp_depayload_handle_sink_event),
(gst_base_rtp_depayload_push_full),
(gst_base_rtp_depayload_set_gst_timestamp),
(gst_base_rtp_depayload_change_state):
Remove code to deal with RTP to GST time conversion, we now just copy
the GST timestamp we receive to the outgoing buffers.
Handle segment and flushes correctly.
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
When we have no valid input timestamp, use the previous rtp timestamp on
the outgoing RTP packet instead of the RTP base time.
2007-09-16 19:31:06 +00:00
Wim Taymans
5dc2aebdee gst-libs/gst/rtp/gstbasertppayload.c: Add some debug info when negotiating caps.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_set_outcaps), (gst_basertppayload_push):
Add some debug info when negotiating caps.
2007-09-15 18:41:27 +00:00
Wim Taymans
a884abc111 gst-libs/gst/rtp/gstrtpbuffer.c: A buffer with an empty payload is also a valid buffer.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data):
A buffer with an empty payload is also a valid buffer.
2007-09-15 00:29:11 +00:00
Wim Taymans
06ded6258e gst-libs/gst/rtp/gstbasertppayload.c: Make sure we start our RTP timestamp from the random base RTP timestamp even if...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_event),
(gst_basertppayload_set_outcaps), (gst_basertppayload_push),
(gst_basertppayload_change_state):
Make sure we start our RTP timestamp from the random base RTP
timestamp even if the buffer timestamp starts from some random value.
2007-09-14 20:52:00 +00:00
Wim Taymans
4764e6044f gst-libs/gst/audio/gstbaseaudiosink.c: Disable pull mode scheduling, we're not ready for it yet and it subtly breaks ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_init):
Disable pull mode scheduling, we're not ready for it yet and it subtly
breaks a lot of things.
2007-09-13 22:52:09 +00:00
Peter Kjellerstedt
3b6df87810 gst/: Printf format fixes (#476128).
Original commit message from CVS:
Patch by: Peter Kjellerstedt  <pkj at axis com>
* gst-libs/gst/app/gstappsink.c:
* gst/flv/gstflvdemux.c:
* gst/flv/gstflvparse.c:
* gst/interleave/deinterleave.c:
* gst/switch/gstswitch.c:
Printf format fixes (#476128).
2007-09-12 08:38:21 +00:00
Tommi Myöhänen
840c5cd805 gst-libs/gst/rtsp/gstrtspconnection.c: Make sure we can not cancel in the middle of receiving a message.
Original commit message from CVS:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_read_internal), (gst_rtsp_connection_read),
(read_body), (gst_rtsp_connection_receive):
Make sure we can not cancel in the middle of receiving a message.
Fixes #475731.
2007-09-11 19:07:57 +00:00
Wim Taymans
c942252430 gst-libs/gst/audio/gstbaseaudiosrc.*: Allow othe clocks than the internal clock to be used for the pipeline.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init), (gst_base_audio_src_init),
(gst_base_audio_src_provide_clock),
(gst_base_audio_src_set_property),
(gst_base_audio_src_get_property), (gst_base_audio_src_create):
* gst-libs/gst/audio/gstbaseaudiosrc.h:
Allow othe clocks than the internal clock to be used for the pipeline.
Add property to disable clock provide.
API: GstBaseAudioSrc::provide-clock
2007-09-10 22:10:54 +00:00
Tim-Philipp Müller
03992b8779 Fix gst_rtp_buffer_set_csrc() and enable csrc-related unit test checks.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtpbuffer.c:
* tests/check/libs/rtp.c:
Fix gst_rtp_buffer_set_csrc() and enable csrc-related unit test checks.
2007-09-07 17:37:03 +00:00
Haakon Sporsheim
b2948f2453 gst-libs/gst/rtp/gstrtpbuffer.c: Fix up GstRTPHeader helper struct so that compilers will not under any circumstances...
Original commit message from CVS:
Based on patch by: Haakon Sporsheim  <haakon.sporsheim at tandberg com>
* gst-libs/gst/rtp/gstrtpbuffer.c:
Fix up GstRTPHeader helper struct so that compilers will not under
any circumstances add padding in between our fields, as currently
happens with MSVC on win32, because that would lead to us sending
out RTP payloads with broken RTP headers (#471194).
Fix assertion guards for gst_rtp_buffer_get_csrc() and _set_csrc().
* tests/check/Makefile.am:
* tests/check/libs/.cvsignore:
* tests/check/libs/rtp.c:
Add some simple unit tests for GstRTPBuffer. Some are disabled
because the code tested still needs fixing (set_csrc() does not work).
2007-09-07 16:46:05 +00:00
Sebastian Dröge
9022ca9d1b Add libgstfft, a FFT library based on Kiss FFT which is
Original commit message from CVS:
Reviewed by:  Stefan Kost  <ensonic@users.sf.net>
* configure.ac:
* gst-libs/gst/Makefile.am:
* gst-libs/gst/fft/Makefile.am:
* 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/gstfft.c: (gst_fft_next_fast_length):
* gst-libs/gst/fft/gstfft.h:
* gst-libs/gst/fft/gstfftf32.c: (gst_fft_f32_new),
(gst_fft_f32_fft), (gst_fft_f32_inverse_fft), (gst_fft_f32_free),
(gst_fft_f32_window), (gst_fft_f32_magnitude), (gst_fft_f32_phase):
* gst-libs/gst/fft/gstfftf32.h:
* gst-libs/gst/fft/gstfftf64.c: (gst_fft_f64_new),
(gst_fft_f64_fft), (gst_fft_f64_inverse_fft), (gst_fft_f64_free),
(gst_fft_f64_window), (gst_fft_f64_magnitude), (gst_fft_f64_phase):
* gst-libs/gst/fft/gstfftf64.h:
* gst-libs/gst/fft/gstffts16.c: (gst_fft_s16_new),
(gst_fft_s16_fft), (gst_fft_s16_inverse_fft), (gst_fft_s16_free),
(gst_fft_s16_window), (gst_fft_s16_magnitude), (gst_fft_s16_phase):
* gst-libs/gst/fft/gstffts16.h:
* gst-libs/gst/fft/gstffts32.c: (gst_fft_s32_new),
(gst_fft_s32_fft), (gst_fft_s32_inverse_fft), (gst_fft_s32_free),
(gst_fft_s32_window), (gst_fft_s32_magnitude), (gst_fft_s32_phase):
* gst-libs/gst/fft/gstffts32.h:
* gst-libs/gst/fft/kiss_fft_f32.c: (kf_bfly2), (kf_bfly4),
(kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
(kiss_fft_f32_alloc), (kiss_fft_f32_stride), (kiss_fft_f32),
(kiss_fft_f32_cleanup), (kiss_fft_f32_next_fast_size):
* gst-libs/gst/fft/kiss_fft_f32.h:
* gst-libs/gst/fft/kiss_fft_f64.c: (kf_bfly2), (kf_bfly4),
(kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
(kiss_fft_f64_alloc), (kiss_fft_f64_stride), (kiss_fft_f64),
(kiss_fft_f64_cleanup), (kiss_fft_f64_next_fast_size):
* gst-libs/gst/fft/kiss_fft_f64.h:
* gst-libs/gst/fft/kiss_fft_s16.c: (kf_bfly2), (kf_bfly4),
(kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
(kiss_fft_s16_alloc), (kiss_fft_s16_stride), (kiss_fft_s16),
(kiss_fft_s16_cleanup), (kiss_fft_s16_next_fast_size):
* gst-libs/gst/fft/kiss_fft_s16.h:
* gst-libs/gst/fft/kiss_fft_s32.c: (kf_bfly2), (kf_bfly4),
(kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
(kiss_fft_s32_alloc), (kiss_fft_s32_stride), (kiss_fft_s32),
(kiss_fft_s32_cleanup), (kiss_fft_s32_next_fast_size):
* gst-libs/gst/fft/kiss_fft_s32.h:
* gst-libs/gst/fft/kiss_fftr_f32.c: (kiss_fftr_f32_alloc),
(kiss_fftr_f32), (kiss_fftri_f32):
* gst-libs/gst/fft/kiss_fftr_f32.h:
* gst-libs/gst/fft/kiss_fftr_f64.c: (kiss_fftr_f64_alloc),
(kiss_fftr_f64), (kiss_fftri_f64):
* gst-libs/gst/fft/kiss_fftr_f64.h:
* gst-libs/gst/fft/kiss_fftr_s16.c: (kiss_fftr_s16_alloc),
(kiss_fftr_s16), (kiss_fftri_s16):
* gst-libs/gst/fft/kiss_fftr_s16.h:
* gst-libs/gst/fft/kiss_fftr_s32.c: (kiss_fftr_s32_alloc),
(kiss_fftr_s32), (kiss_fftri_s32):
* gst-libs/gst/fft/kiss_fftr_s32.h:
* gst-libs/gst/fft/kiss_version:
* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
* pkgconfig/gstreamer-plugins-base.pc.in:
Add libgstfft, a FFT library based on Kiss FFT which is
BSD licensed. Supported sample formats are int16, int32,
float and double. For those formats a real FFT and IFFT
can be done, different windowing functions can be applied
and functions for extracting the magnitude and phase exist.
Fixes #468619.
* docs/libs/Makefile.am:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
Integrate libgstfft into the docs.
* tests/check/Makefile.am:
* tests/check/libs/fft.c: (GST_START_TEST), (fft_suite), (main):
Add unit tests for libgstfft, currently only testing the FFT.
Unit tests for IFFT will follow soon.
2007-09-06 07:00:36 +00:00
Peter Kjellerstedt
f149134834 gst-libs/gst/sdp/gstsdpmessage.*: Separate INIT_ARRAY() and related macros into two versions, one for structures and ...
Original commit message from CVS:
Patch by: Peter Kjellerstedt  <pkj at axis com>
* gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_time_init),
(gst_sdp_message_init), (gst_sdp_message_uninit),
(is_multicast_address), (gst_sdp_message_as_text),
(gst_sdp_message_get_origin), (gst_sdp_message_set_connection),
(gst_sdp_message_get_connection), (gst_sdp_message_add_bandwidth),
(gst_sdp_message_add_time), (gst_sdp_message_add_zone),
(gst_sdp_message_get_key), (gst_sdp_message_get_attribute_val_n),
(gst_sdp_message_get_attribute_val), (gst_sdp_message_add_media),
(gst_sdp_media_init), (gst_sdp_media_uninit),
(gst_sdp_media_as_text), (gst_sdp_media_set_port_info),
(gst_sdp_media_connections_len), (gst_sdp_media_add_connection),
(gst_sdp_media_bandwidths_len), (gst_sdp_media_get_bandwidth),
(gst_sdp_media_add_bandwidth), (gst_sdp_media_attributes_len),
(gst_sdp_parse_line), (print_media), (gst_sdp_message_dump):
* gst-libs/gst/sdp/gstsdpmessage.h:
Separate INIT_ARRAY() and related macros into two versions, one for
structures and one for pointers (e.g., INIT_ARRAY() and
INIT_PTR_ARRAY()). This fixes a segmentation error on freeing the
lists of emails and phone numbers.
Add missing const as appropriate.
Change all gint to guint since they all actually represent unsigned
values.
Do not use time as a variable name as it shadows the global time().
Add gst_sdp_message_as_text() and gst_sdp_media_as_text().
Actually implement gst_sdp_message_add_time().
Make gst_sdp_message_add_time() take repeat times as an argument.
Store repeat times in GstSDPTime as a GArray rather than as gchar**.
Corrected the definition of gst_sdp_media_get_bandwidth() (was
misspelled as badwidth).
gst-indented and a little clean up. Fixes #471067.
2007-09-05 23:07:40 +00:00
Tim-Philipp Müller
7ea0798a9c gst-libs/gst/rtp/gstrtpbuffer.c: Need to include stdlib.h for abs() here too.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtpbuffer.c:
Need to include stdlib.h for abs() here too.
2007-09-05 14:09:15 +00:00
Wim Taymans
56e39e7c1c gst-libs/gst/rtp/gstbasertpaudiopayload.c: Return FALSE from the event handler to let the parent class handle the event.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_payload_audio_handle_event):
Return FALSE from the event handler to let the parent class handle the
event.
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_chain), (gst_base_rtp_depayload_push_full):
Mark outgoing buffers as DISCONT if the incomming buffer was DISCONT.
* gst-libs/gst/rtp/gstbasertppayload.c:
Bump the MTU to 1400.
2007-09-04 16:18:48 +00:00
Wim Taymans
6f93db5ab5 Fix parsing of RB blocks.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_get_rb),
(gst_rtcp_packet_sdes_copy_entry), (gst_rtcp_ntp_to_unix),
(gst_rtcp_unix_to_ntp):
* gst-libs/gst/rtp/gstrtcpbuffer.h:
Fix parsing of RB blocks.
Fix docs.
Added helper functions to convert to/from UNIX and NTP time.
API: gst_rtcp_ntp_to_unix()
API: gst_rtcp_unix_to_ntp()
* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data),
(gst_rtp_buffer_get_header_len),
(gst_rtp_buffer_get_extension_data),
(gst_rtp_buffer_get_payload_subbuffer),
(gst_rtp_buffer_get_payload_len), (gst_rtp_buffer_get_payload),
(gst_rtp_buffer_ext_timestamp):
* gst-libs/gst/rtp/gstrtpbuffer.h:
Fix some more docs.
Implement handling of packets with extensions.
Fix padding check in _validate().
Added function to get extension data.
API: gst_rtp_buffer_get_header_len()
API: gst_rtp_buffer_get_extension_data()
2007-09-03 19:31:11 +00:00
Wim Taymans
0cfb3152b9 gst-libs/gst/rtp/gstbasertpdepayload.c: Add some more docs for the queue-delay property and fix a typo in a comment.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_class_init),
(gst_base_rtp_depayload_set_gst_timestamp):
Add some more docs for the queue-delay property and fix a typo in a
comment.
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
Fix typo.
2007-09-03 19:19:35 +00:00
Wim Taymans
c2460052b3 gst-libs/gst/audio/gstbaseaudiosink.c: When skew slaving, try to hover around the middle of a segment so that we at m...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_skew_slaving), (gst_base_audio_sink_render),
(gst_base_audio_sink_change_state):
When skew slaving, try to hover around the middle of a segment so that
we at most drift by half a segment.
If we are aligning in the oposite direction of the clock skew, we don't
have to resync.
2007-09-03 19:17:33 +00:00
Wim Taymans
210100078d gst-libs/gst/rtp/gstbasertpdepayload.c: Be less silly with the segment start, just apply the clock-base to the timest...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_setcaps),
(gst_base_rtp_depayload_set_gst_timestamp):
Be less silly with the segment start, just apply the clock-base to the
timestamp.
2007-08-31 21:07:20 +00:00
Wim Taymans
827967c8e8 gst-libs/gst/rtp/gstbasertpdepayload.*: Deprecate the queue handling thread thing and remove the code.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_class_init),
(gst_base_rtp_depayload_finalize),
(gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
(gst_base_rtp_depayload_handle_sink_event),
(gst_base_rtp_depayload_set_gst_timestamp),
(gst_base_rtp_depayload_change_state):
* gst-libs/gst/rtp/gstbasertpdepayload.h:
Deprecate the queue handling thread thing and remove the code.
Use new method to calculate the extended timestamp.
2007-08-31 15:58:30 +00:00
Wim Taymans
27ea51ec37 gst-libs/gst/rtp/gstrtcpbuffer.c: Use g_strndup which does exactly what we want.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtcpbuffer.c:
(gst_rtcp_packet_sdes_copy_entry):
Use g_strndup which does exactly what we want.
* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_compare_seqnum),
(gst_rtp_buffer_ext_timestamp):
* gst-libs/gst/rtp/gstrtpbuffer.h:
Add helper function to compare seqnums.
Add helper function to calculate extended timestamps.
API: gst_rtp_buffer_compare_seqnum()
API: gst_rtp_buffer_ext_timestamp()
2007-08-31 15:21:13 +00:00
Wim Taymans
fdc42d47b4 gst-libs/gst/rtp/gstrtcpbuffer.*: Fix and document SDES item data function.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtcpbuffer.c:
(gst_rtcp_packet_sdes_get_entry),
(gst_rtcp_packet_sdes_copy_entry):
* gst-libs/gst/rtp/gstrtcpbuffer.h:
Fix and document SDES item data function.
Add new function that makes a proper copy of SDES item data.
API: gst_rtcp_packet_sdes_copy_entry()
2007-08-30 21:59:23 +00:00
Tim-Philipp Müller
b8f1da91d1 API: also add gst_install_plugins_supported() while we're at it (see #470456).
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/pbutils/install-plugins.c:
* gst-libs/gst/pbutils/install-plugins.h:
* tests/check/libs/pbutils.c:
API: also add gst_install_plugins_supported() while we're at it
(see #470456).
2007-08-28 14:58:17 +00:00
Tim-Philipp Müller
f344ec6b8a API: add gst_missing_*_installer_detail_new() convenience API so that applications that know exactly what they're mis...
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/pbutils/missing-plugins.c:
* gst-libs/gst/pbutils/missing-plugins.h:
* tests/check/libs/pbutils.c:
API: add gst_missing_*_installer_detail_new() convenience API so
that applications that know exactly what they're missing can request
installer detail strings for those items directly instead of having
to first create a dummy missing-plugin message and then get the
installer detail string from that.  Fixes #470456.
2007-08-28 14:23:55 +00:00
Tim-Philipp Müller
e2dbf33a7c gst-libs/gst/pbutils/missing-plugins.c: Add missing separator in PID fallback case.
Original commit message from CVS:
* gst-libs/gst/pbutils/missing-plugins.c:
(gst_missing_plugin_message_get_installer_detail):
Add missing separator in PID fallback case.
2007-08-26 14:14:33 +00:00
Stefan Kost
1772d04dda Revert unwanted commit. many thanks to moap. I want a fix for https://thomas.apestaart.org/moap/trac/ticket/239
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:
* ext/gnomevfs/gstgnomevfssrc.c:
* ext/gnomevfs/gstgnomevfssrc.h:
* gst-libs/gst/Makefile.am:
* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
* pkgconfig/gstreamer-plugins-base.pc.in:
* sys/v4l/v4lsrc_calls.c:
* tests/examples/Makefile.am:
* win32/common/config.h:
Revert unwanted commit. many thanks to moap. I want a fix for
https://thomas.apestaart.org/moap/trac/ticket/239
2007-08-23 10:58:42 +00:00
Stefan Kost
a5e777fac3 Original commit message from CVS:
reviewed by: <delete if not using a buddy>
patch by: <delete if not someone else's patch>
* configure.ac:
* docs/libs/Makefile.am:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* ext/gnomevfs/gstgnomevfssrc.c:
* ext/gnomevfs/gstgnomevfssrc.h:
* gst-libs/gst/Makefile.am:
* gst-libs/gst/audio/gstaudiofilter.h:
* gst/typefind/gsttypefindfunctions.c:
* gst/volume/gstvolume.c:
* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
* pkgconfig/gstreamer-plugins-base.pc.in:
* sys/v4l/v4lsrc_calls.c:
* tests/examples/Makefile.am:
* win32/common/config.h:
2007-08-23 08:33:43 +00:00
Wim Taymans
478a6592de gst-libs/gst/audio/audio.c: Clarify the docs a little.
Original commit message from CVS:
* gst-libs/gst/audio/audio.c:
Clarify the docs a little.
2007-08-22 15:29:04 +00:00
Sebastian Dröge
846ddaa550 gst-libs/gst/audio/audio.h: Use gst_util_uint64_scale() instead of doing the math with double for GST_FRAMES_TO_CLOCK...
Original commit message from CVS:
* gst-libs/gst/audio/audio.h:
Use gst_util_uint64_scale() instead of doing the math
with double for GST_FRAMES_TO_CLOCK_TIME() and
GST_CLOCK_TIME_TO_FRAMES(). For large timestamps this
prevents rounding errors. Fixes #467667.
2007-08-17 15:24:43 +00:00
Wim Taymans
01d9553d43 gst-libs/gst/rtsp/gstrtspconnection.*: Small cleanups.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_connect), (gst_rtsp_connection_write),
(gst_rtsp_connection_read), (gst_rtsp_connection_poll):
* gst-libs/gst/rtsp/gstrtspconnection.h:
Small cleanups.
On shutdown, don't read the control socket yet.
Set timeout value correctly in all cases.
Add function to check if the server accepts reads or writes.
API: gst_rtsp_connection_poll()
* gst-libs/gst/rtsp/gstrtspdefs.h:
Fix compilation with -pedantic.
Add enum for _poll.
2007-08-17 13:42:49 +00:00
Wim Taymans
c17a721e0a gst-libs/gst/app/gstappsink.c: Override the preroll vmethod instead of overriding the render method twice.
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init):
Override the preroll vmethod instead of overriding the render method
twice.
2007-08-16 17:11:48 +00:00
Olivier Crete
b78030f77d gst-libs/gst/rtp/gstbasertppayload.*: Add getcaps vfunc to basertppayload. See #465146.
Original commit message from CVS:
Patch by: Olivier Crete  <tester at tester ca>
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
(gst_basertppayload_getcaps):
* gst-libs/gst/rtp/gstbasertppayload.h:
Add getcaps vfunc to basertppayload. See #465146.
2007-08-16 16:06:21 +00:00
Tim-Philipp Müller
0afe67c9e0 gst-libs/gst/pbutils/: Small docs fix and addition.
Original commit message from CVS:
* gst-libs/gst/pbutils/install-plugins.c:
* gst-libs/gst/pbutils/missing-plugins.c:
Small docs fix and addition.
2007-08-15 17:05:45 +00:00
Wim Taymans
1ec11dbc8e gst-libs/gst/app/gstappsink.c: Don't use new API.
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_flush_unlocked):
Don't use new API.
2007-08-14 17:47:34 +00:00
Wim Taymans
dd72f88a8c gst-libs/gst/app/gstappsink.*: Make love to appsink.
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_base_init),
(gst_app_sink_class_init), (gst_app_sink_dispose),
(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_get_caps),
(gst_app_sink_set_caps), (gst_app_sink_end_of_stream),
(gst_app_sink_pull_preroll), (gst_app_sink_pull_buffer):
* gst-libs/gst/app/gstappsink.h:
Make love to appsink.
Make it support pulling of the preroll buffer.
Add docs and debug statements.
Fix some races wrt to EOS handling and stopping.
Implement getcaps.
Implement FLUSHING.
API: gst_app_sink_pull_preroll()
2007-08-14 17:38:05 +00:00
Wim Taymans
3b7071a16f gst-libs/gst/rtp/gstbasertppayload.*: Improve caps negotiation so that downstream elements can confiure certain RTP p...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_set_outcaps):
* gst-libs/gst/rtp/gstbasertppayload.h:
Improve caps negotiation so that downstream elements can confiure
certain RTP properties by fixing them on the caps. See #465146.
Add docs.
2007-08-12 16:30:36 +00:00
Tim-Philipp Müller
2d5d5ac891 Mark as deprecated some macros which were presumably meant to be private API and accidentally exposed in the public h...
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
* gst-libs/gst/rtp/gstbasertpdepayload.h:
Mark as deprecated some macros which were presumably meant to be
private API and accidentally exposed in the public header file.
Also actually _init() lock (only works at the moment because the
struct is zeroed out when created and the initial values in the
mutex struct are zeroes too). (#459585)
2007-08-11 12:39:51 +00:00
Damien Lespiau
9b8c837165 Fix compilation on windows. Fixes #464320.
Original commit message from CVS:
Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
* configure.ac:
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_connect):
Fix compilation on windows. Fixes #464320.
2007-08-07 15:13:46 +00:00
Wim Taymans
607fa48ad8 gst-libs/gst/rtsp/gstrtsptransport.c: Add rdt manager for rdt transport.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtsptransport.c: (gst_rtsp_transport_parse):
Add rdt manager for rdt transport.
Fix parsing of RDT transport.
2007-08-03 15:44:01 +00:00
Jan Schmidt
d5dc054ea3 gst-libs/gst/audio/audio.c: When clipping a buffer with no timestamp, assume it is within the segment without warnings.
Original commit message from CVS:
* gst-libs/gst/audio/audio.c:
When clipping a buffer with no timestamp, assume it is
within the segment without warnings.
Fixes: #460978
2007-07-27 17:10:47 +00:00
Wim Taymans
be5ef4b0ad gst-libs/gst/rtsp/gstrtspextension.c: Fire the signal on the object, not the interface.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspextension.c: (gst_rtsp_extension_send):
Fire the signal on the object, not the interface.
2007-07-27 11:16:23 +00:00
Jan Schmidt
1846b1a84d gst-libs/gst/rtsp/.cvsignore: Ber. Don't include the full path, idiot.
Original commit message from CVS:
* gst-libs/gst/rtsp/.cvsignore:
Ber. Don't include the full path, idiot.
2007-07-27 09:17:19 +00:00
Jan Schmidt
c339ca80c3 gst-libs/gst/rtsp/.cvsignore: Ignore generated files.
Original commit message from CVS:
* gst-libs/gst/rtsp/.cvsignore:
Ignore generated files.
2007-07-27 08:29:29 +00:00
Jan Schmidt
aa14635c47 gst-libs/gst/: Move the rtspextension.h interface into gstrtspextension.h as part of libgstrtsp instead of libgstinte...
Original commit message from CVS:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/interfaces/interfaces-marshal.list:
* gst-libs/gst/interfaces/rtspextension.c:
* gst-libs/gst/interfaces/rtspextension.h:
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/rtsp/gstrtsp.h:
* gst-libs/gst/rtsp/gstrtspextension.c:
(gst_rtsp_extension_get_type), (gst_rtsp_extension_iface_init),
(gst_rtsp_extension_detect_server),
(gst_rtsp_extension_before_send), (gst_rtsp_extension_after_send),
(gst_rtsp_extension_parse_sdp), (gst_rtsp_extension_setup_media),
(gst_rtsp_extension_configure_stream),
(gst_rtsp_extension_get_transports),
(gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
* gst-libs/gst/rtsp/gstrtspextension.h:
* gst-libs/gst/rtsp/rtsp-marshal.list:
Move the rtspextension.h interface into gstrtspextension.h
as part of libgstrtsp instead of libgstinterfaces, because it's
only for use within plugins, not applications.
Add stuff to do the enum & marshal generation needed in libgstrtsp now.
Use the GST_TYPE_RTSP_RESULT enum type for the return value of the
signal that the GstRTSPExtension interface emits, since G_TYPE_ENUM
is abstract.
2007-07-26 19:57:15 +00:00
Wim Taymans
6d1a34eff2 gst-libs/gst/interfaces/: Fix marshaller for the send signal.
Original commit message from CVS:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/interfaces/interfaces-marshal.list:
* gst-libs/gst/interfaces/rtspextension.c:
(gst_rtsp_extension_iface_init),
(gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
* gst-libs/gst/interfaces/rtspextension.h:
Fix marshaller for the send signal.
Add URL to stream selection interface method.
2007-07-26 15:48:01 +00:00
Jan Schmidt
50a3a239a0 gst-libs/gst/riff/Makefile.am: Pull in our dependencies from -base before those from outside.
Original commit message from CVS:
* gst-libs/gst/riff/Makefile.am:
Pull in our dependencies from -base before those from outside.
2007-07-26 15:35:43 +00:00
Wim Taymans
2c35823bdf API: gst_rtsp_base64_decode_ip()
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_decode_ip):
* gst-libs/gst/rtsp/gstrtspbase64.h:
API: gst_rtsp_base64_decode_ip()
Added function to decode Base64 in-place.
2007-07-26 14:33:01 +00:00
Wim Taymans
8db50d49f7 gst-libs/gst/sdp/gstsdpmessage.*: Constify args where we can.
Original commit message from CVS:
* gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_media_get_media),
(gst_sdp_media_get_port), (gst_sdp_media_get_num_ports),
(gst_sdp_media_get_proto), (gst_sdp_media_formats_len),
(gst_sdp_media_get_format), (gst_sdp_media_get_information),
(gst_sdp_media_connections_len), (gst_sdp_media_get_connection),
(gst_sdp_media_bandwidths_len), (gst_sdp_media_get_badwidth),
(gst_sdp_media_get_key), (gst_sdp_media_attributes_len),
(gst_sdp_media_get_attribute), (gst_sdp_media_get_attribute_val_n),
(gst_sdp_media_get_attribute_val):
* gst-libs/gst/sdp/gstsdpmessage.h:
Constify args where we can.
2007-07-25 18:20:36 +00:00
Wim Taymans
256d005e49 gst-libs/gst/interfaces/: Move interface for RTSP extensions from -good to here.
Original commit message from CVS:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/interfaces/rtspextension.c:
(gst_rtsp_extension_get_type), (gst_rtsp_extension_iface_init),
(gst_rtsp_extension_detect_server),
(gst_rtsp_extension_before_send), (gst_rtsp_extension_after_send),
(gst_rtsp_extension_parse_sdp), (gst_rtsp_extension_setup_media),
(gst_rtsp_extension_configure_stream),
(gst_rtsp_extension_get_transports),
(gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
* gst-libs/gst/interfaces/rtspextension.h:
Move interface for RTSP extensions from -good to here.
Added helper methods to invoke interface methods.
2007-07-25 18:18:49 +00:00
Wim Taymans
77c284a31f Fix some more RTSP docs.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtsp/gstrtspdefs.h:
* gst-libs/gst/rtsp/gstrtspmessage.c: (key_value_foreach),
(gst_rtsp_message_get_type), (gst_rtsp_message_parse_request),
(gst_rtsp_message_init_response),
(gst_rtsp_message_parse_response), (gst_rtsp_message_new_data),
(gst_rtsp_message_parse_data), (gst_rtsp_message_add_header),
(gst_rtsp_message_remove_header), (gst_rtsp_message_get_header),
(gst_rtsp_message_append_headers), (gst_rtsp_message_set_body),
(gst_rtsp_message_get_body), (dump_key_value):
* gst-libs/gst/rtsp/gstrtspmessage.h:
* gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
(parse_npt_range), (parse_clock_range), (parse_smpte_range),
(gst_rtsp_range_parse):
* gst-libs/gst/rtsp/gstrtsprange.h:
* gst-libs/gst/rtsp/gstrtsptransport.c:
* gst-libs/gst/rtsp/gstrtspurl.c:
Fix some more RTSP docs.
Add some missing methods for dealing with messages.
2007-07-25 11:22:30 +00:00
Wim Taymans
3dff14d6b1 Added beginnings of RTSP documentation.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_encode):
* gst-libs/gst/rtsp/gstrtspbase64.h:
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_connect), (add_auth_header),
(gst_rtsp_connection_write), (gst_rtsp_connection_send),
(read_body), (gst_rtsp_connection_receive),
(gst_rtsp_connection_next_timeout),
(gst_rtsp_connection_reset_timeout),
(gst_rtsp_connection_set_auth):
* gst-libs/gst/rtsp/gstrtspconnection.h:
* gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status):
* gst-libs/gst/rtsp/gstrtspdefs.h:
* gst-libs/gst/rtsp/gstrtspmessage.h:
* gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
(parse_npt_range), (parse_clock_range), (parse_smpte_range),
(gst_rtsp_range_parse):
* gst-libs/gst/rtsp/gstrtspurl.h:
Added beginnings of RTSP documentation.
2007-07-24 19:19:33 +00:00
Wim Taymans
ee42361c89 Document the SDP library.
Original commit message from CVS:
* docs/libs/Makefile.am:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/sdp/gstsdp.h:
* gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_message_set_origin),
(gst_sdp_message_set_connection), (gst_sdp_message_add_bandwidth),
(gst_sdp_message_add_time), (gst_sdp_message_add_zone),
(gst_sdp_message_set_key), (gst_sdp_message_get_attribute_val_n),
(gst_sdp_message_get_attribute_val),
(gst_sdp_message_add_attribute), (gst_sdp_media_new),
(gst_sdp_media_init), (gst_sdp_media_uninit), (gst_sdp_media_free),
(gst_sdp_media_get_media), (gst_sdp_media_set_media),
(gst_sdp_media_get_port), (gst_sdp_media_get_num_ports),
(gst_sdp_media_set_port_info), (gst_sdp_media_get_proto),
(gst_sdp_media_set_proto), (gst_sdp_media_formats_len),
(gst_sdp_media_get_format), (gst_sdp_media_add_format),
(gst_sdp_media_get_information), (gst_sdp_media_set_information),
(gst_sdp_media_connections_len), (gst_sdp_media_get_connection),
(gst_sdp_media_add_connection), (gst_sdp_media_bandwidths_len),
(gst_sdp_media_get_badwidth), (gst_sdp_media_add_bandwidth),
(gst_sdp_media_set_key), (gst_sdp_media_get_key),
(gst_sdp_media_attributes_len), (gst_sdp_media_add_attribute),
(gst_sdp_media_get_attribute_val_n),
(gst_sdp_media_get_attribute_val), (gst_sdp_message_parse_buffer),
(print_media), (gst_sdp_message_dump):
* gst-libs/gst/sdp/gstsdpmessage.h:
Document the SDP library.
Add some of the missing SDPMedia methods.
2007-07-24 17:37:03 +00:00
Wim Taymans
19e0dd3140 Move SDP and RTSP from helper objects in -good to a reusable library.
Original commit message from CVS:
* configure.ac:
* gst-libs/gst/Makefile.am:
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_encode):
* gst-libs/gst/rtsp/gstrtspbase64.h:
* gst-libs/gst/rtsp/gstrtspconnection.c: (inet_aton),
(gst_rtsp_connection_create), (gst_rtsp_connection_connect),
(add_auth_header), (add_date_header), (gst_rtsp_connection_write),
(gst_rtsp_connection_send), (read_line), (read_string), (read_key),
(parse_response_status), (parse_request_line), (parse_line),
(gst_rtsp_connection_read), (read_body),
(gst_rtsp_connection_receive), (gst_rtsp_connection_close),
(gst_rtsp_connection_free), (gst_rtsp_connection_next_timeout),
(gst_rtsp_connection_reset_timeout), (gst_rtsp_connection_flush),
(gst_rtsp_connection_set_auth):
* gst-libs/gst/rtsp/gstrtspconnection.h:
* gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status),
(gst_rtsp_strresult), (gst_rtsp_method_as_text),
(gst_rtsp_version_as_text), (gst_rtsp_header_as_text),
(gst_rtsp_status_as_text), (gst_rtsp_find_header_field),
(gst_rtsp_find_method):
* gst-libs/gst/rtsp/gstrtspdefs.h:
* gst-libs/gst/rtsp/gstrtspmessage.c: (key_value_foreach),
(gst_rtsp_message_new), (gst_rtsp_message_init),
(gst_rtsp_message_new_request), (gst_rtsp_message_init_request),
(gst_rtsp_message_new_response), (gst_rtsp_message_init_response),
(gst_rtsp_message_init_data), (gst_rtsp_message_unset),
(gst_rtsp_message_free), (gst_rtsp_message_add_header),
(gst_rtsp_message_remove_header), (gst_rtsp_message_get_header),
(gst_rtsp_message_append_headers), (gst_rtsp_message_set_body),
(gst_rtsp_message_take_body), (gst_rtsp_message_get_body),
(gst_rtsp_message_steal_body), (dump_mem), (dump_key_value),
(gst_rtsp_message_dump):
* gst-libs/gst/rtsp/gstrtspmessage.h:
* gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
(parse_npt_range), (parse_clock_range), (parse_smpte_range),
(gst_rtsp_range_parse), (gst_rtsp_range_free):
* gst-libs/gst/rtsp/gstrtsprange.h:
* gst-libs/gst/rtsp/gstrtsptransport.c: (gst_rtsp_transport_new),
(gst_rtsp_transport_init), (gst_rtsp_transport_get_mime),
(gst_rtsp_transport_get_manager), (parse_mode), (parse_range),
(range_as_text), (rtsp_transport_mode_as_text),
(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
(gst_rtsp_transport_parse), (gst_rtsp_transport_as_text),
(gst_rtsp_transport_free):
* gst-libs/gst/rtsp/gstrtsptransport.h:
* gst-libs/gst/rtsp/gstrtspurl.c: (gst_rtsp_url_parse),
(gst_rtsp_url_free), (gst_rtsp_url_set_port),
(gst_rtsp_url_get_port), (gst_rtsp_url_get_request_uri):
* gst-libs/gst/rtsp/gstrtspurl.h:
* gst-libs/gst/sdp/Makefile.am:
* gst-libs/gst/sdp/gstsdp.h:
* gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_origin_init),
(gst_sdp_connection_init), (gst_sdp_bandwidth_init),
(gst_sdp_time_init), (gst_sdp_zone_init), (gst_sdp_key_init),
(gst_sdp_attribute_init), (gst_sdp_message_new),
(gst_sdp_message_init), (gst_sdp_message_uninit),
(gst_sdp_message_free), (gst_sdp_media_new), (gst_sdp_media_init),
(gst_sdp_media_uninit), (gst_sdp_media_free),
(gst_sdp_message_set_origin), (gst_sdp_message_get_origin),
(gst_sdp_message_set_connection), (gst_sdp_message_get_connection),
(gst_sdp_message_add_bandwidth), (gst_sdp_message_add_time),
(gst_sdp_message_add_zone), (gst_sdp_message_set_key),
(gst_sdp_message_get_key), (gst_sdp_message_get_attribute_val_n),
(gst_sdp_message_get_attribute_val),
(gst_sdp_message_add_attribute), (gst_sdp_message_add_media),
(gst_sdp_media_add_attribute), (gst_sdp_media_add_bandwidth),
(gst_sdp_media_add_format), (gst_sdp_media_get_attribute),
(gst_sdp_media_get_attribute_val_n),
(gst_sdp_media_get_attribute_val), (gst_sdp_media_get_format),
(read_string), (read_string_del), (gst_sdp_parse_line),
(gst_sdp_message_parse_buffer), (print_media),
(gst_sdp_message_dump):
* gst-libs/gst/sdp/gstsdpmessage.h:
* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
Move SDP and RTSP from helper objects in -good to a reusable library.
Use a proper gst_ namespace.
2007-07-24 11:52:56 +00:00
Sebastian Dröge
6be2524031 API: Add buffer clipping function for raw audio buffers. Fixes #456656.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/audio.c: (gst_audio_buffer_clip):
* gst-libs/gst/audio/audio.h:
* tests/check/libs/audio.c: (GST_START_TEST), (audio_suite):
API: Add buffer clipping function for raw audio buffers. Fixes #456656.
Also add deprecation guards for gst_audio_structure_set_int() to the
header.
2007-07-23 18:26:09 +00:00
Jan Schmidt
0776d87e32 gst-libs/gst/interfaces/mixer.h: Add padding vars in place of the signal pointers when building with DISABLE_DEPRECAT...
Original commit message from CVS:
* gst-libs/gst/interfaces/mixer.h:
Add padding vars in place of the signal pointers
when building with DISABLE_DEPRECATED so that the
interface structure doesn't change size.
2007-07-21 09:56:09 +00:00
Marc-Andre Lureau
c161e29307 Fixes: #152864
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* ext/alsa/gstalsamixer.c:
* ext/alsa/gstalsamixer.h:
* ext/alsa/gstalsamixerelement.c:
* ext/alsa/gstalsamixertrack.c:
* gst-libs/gst/interfaces/mixer.c:
* gst-libs/gst/interfaces/mixer.h:
* gst-libs/gst/interfaces/mixeroptions.c:
* gst-libs/gst/interfaces/mixeroptions.h:
* gst-libs/gst/interfaces/mixertrack.c:
* gst-libs/gst/interfaces/mixertrack.h:
* tests/check/Makefile.am:
* tests/check/libs/mixer.c:
Patch By: Marc-Andre Lureau <marcandre.lureau@gmail.com>
Fixes: #152864
Add support for notifying mixer changes on the message bus, and
implement it in alsamixer.
API: gst_mixer_get_mixer_flags
API: gst_mixer_message_parse_mute_toggled
API: gst_mixer_message_parse_record_toggled
API: gst_mixer_message_parse_volume_changed
API: gst_mixer_message_parse_option_changed
API: GstMixerMessageType
API: GstMixerFlags
2007-07-21 09:21:12 +00:00
Wim Taymans
d0e9a76a95 gst-libs/gst/rtp/gstbasertppayload.c: Don't break ABI, restore previous ranges. Keep the default random selection of ...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_class_init), (gst_basertppayload_init),
(gst_basertppayload_set_property),
(gst_basertppayload_get_property):
Don't break ABI, restore previous ranges. Keep the default random
selection of timestamp and seqnum offset but as soon as the app sets a
specific value, use that one.
2007-07-16 10:10:28 +00:00
Wim Taymans
c82275a51d gst-libs/gst/rtp/gstbasertppayload.*: Fix ranges of rtp payloader properties so that the full range can be used in ad...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_class_init), (gst_basertppayload_init),
(gst_basertppayload_event), (gst_basertppayload_push),
(gst_basertppayload_set_property),
(gst_basertppayload_get_property),
(gst_basertppayload_change_state):
* gst-libs/gst/rtp/gstbasertppayload.h:
Fix ranges of rtp payloader properties so that the full range can be
used in addition to -1 (random).
Fix wrong seqnum reporting in caps.
Fixes #420326.
2007-07-14 17:23:42 +00:00
Stefan Kost
aac0353ce6 gst-libs/gst/: Make gtk-doc happy.
Original commit message from CVS:
* gst-libs/gst/interfaces/propertyprobe.c:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/tag/gstvorbistag.c:
Make gtk-doc happy.
2007-07-10 20:46:41 +00:00
Tim-Philipp Müller
8a499651b9 gst-libs/gst/audio/gstbaseaudiosink.c: Quick hack to make audiosinks stop at EOS when operating in pull-mode; needs t...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_callback):
Quick hack to make audiosinks stop at EOS when operating in
pull-mode; needs to be fixed properly some day.
2007-07-08 13:07:38 +00:00
Tim-Philipp Müller
28ef3f5ddf gst-libs/gst/tag/tags.c: Don't pass trailing zeroes in fixed-size string arrays in ID3v1 tags to utf8-validate; fixes...
Original commit message from CVS:
* gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
Don't pass trailing zeroes in fixed-size string arrays in ID3v1 tags
to utf8-validate; fixes recognition of ID3v1 tags in UTF-8 encoding
(#451707); also, output some debugging info when dealing with
freeform strings.
* tests/check/libs/tag.c: (GST_START_TEST), (tag_suite):
Add unit test for the above.
2007-06-27 22:30:19 +00:00
Tim-Philipp Müller
f637e3b80c gst-libs/gst/pbutils/descriptions.c: Add description for Windows Media RTP caps.
Original commit message from CVS:
* gst-libs/gst/pbutils/descriptions.c: (caps_are_rtp_caps):
Add description for Windows Media RTP caps.
* gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
Remove RTP fields that don't define the format from caps.
2007-06-27 12:55:20 +00:00
Andy Wingo
ae6fd1b3f2 gst-libs/gst/audio/gstbaseaudiosink.c
Original commit message from CVS:
2007-06-19  Andy Wingo  <wingo@pobox.com>

* gst-libs/gst/audio/gstbaseaudiosink.c
(gst_base_audio_sink_init): Enable pull-mode operation.
2007-06-19 19:13:04 +00:00
Michael Smith
ba06a86e01 gst-libs/gst/riff/riff-media.c: Change minimum rate back to 1000 to allow low-sample-rate wav files to play back.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Change minimum rate back to 1000 to allow low-sample-rate wav files
to play back.
2007-06-19 09:34:35 +00:00
Sébastien Moutte
a6d8c4109e gst-libs/gst/rtp/gstbasertpdepayload.c: Use G_GINT64_CONSTANT macro for int64 constant.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_set_gst_timestamp):
Use G_GINT64_CONSTANT macro for int64 constant.
* win32/common/libgstinterfaces.def:
* win32/common/libgsttag.def:
Add new exported functions.
2007-06-07 21:08:38 +00:00
Tim-Philipp Müller
257a20e77a gst-libs/gst/riff/: Use gst_tag_utf8_from_freeform_string() from libgsttag instead of our own implementation.
Original commit message from CVS:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/riff/riff-read.c: (gst_riff_parse_info):
Use gst_tag_utf8_from_freeform_string() from libgsttag instead of
our own implementation.
2007-06-05 16:20:44 +00:00
Wim Taymans
9dac555993 gst-libs/gst/rtp/gstbasertpdepayload.c: Handle timestamp wraparound.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_setcaps),
(gst_base_rtp_depayload_set_gst_timestamp),
(gst_base_rtp_depayload_change_state):
Handle timestamp wraparound.
2007-06-05 16:19:30 +00:00
Wim Taymans
d51693e960 gst-libs/gst/riff/riff-media.c: In riff, the depth is stored in the size field but it just means that the least signi...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
In riff, the depth is stored in the size field but it just means that
the least significant bits are cleared. We can therefore just play
the sample as if it had a depth == width. Fixes: #440997
Patch by: Wim Taymans <wim@fluendo.com>
Patch by: Sebastian Dröge  <slomo@circular-chaos.org>
2007-05-31 17:08:58 +00:00
Jan Schmidt
d6ef01a879 gst-libs/gst/floatcast/floatcast.h: Define inline when needed on win32 builds. Fixes: #441295
Original commit message from CVS:
* gst-libs/gst/floatcast/floatcast.h:
Define inline when needed on win32 builds. Fixes: #441295
2007-05-31 16:36:22 +00:00
Jan Schmidt
588bc09c33 Revert commits towards #152864 made so far. We'll pick it up again after the 0.10.13 release.
Original commit message from CVS:
* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_ensure_track_list),
(gst_alsa_mixer_free), (gst_alsa_mixer_update),
(gst_alsa_mixer_get_volume), (gst_alsa_mixer_set_volume),
(gst_alsa_mixer_set_mute), (gst_alsa_mixer_set_record),
(gst_alsa_mixer_set_option), (gst_alsa_mixer_get_option):
* ext/alsa/gstalsamixer.h:
* ext/alsa/gstalsamixerelement.c:
(gst_alsa_mixer_element_interface_supported),
(gst_alsa_mixer_element_finalize), (gst_alsa_mixer_element_init),
(gst_alsa_mixer_element_set_property),
(gst_alsa_mixer_element_get_property),
(gst_alsa_mixer_element_change_state):
* ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_update):
* gst-libs/gst/interfaces/mixer.c: (gst_mixer_volume_changed),
(gst_mixer_option_changed):
* gst-libs/gst/interfaces/mixer.h:
Revert commits towards #152864 made so far. We'll pick it up again
after the 0.10.13 release.
2007-05-25 10:07:26 +00:00
Wim Taymans
b2fdf703c9 gst-libs/gst/audio/gstbaseaudiosink.c: After an interrupt (PAUSED/flush) assume that the next sample should not be al...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
After an interrupt (PAUSED/flush) assume that the next sample should not
be aligned to the previous sample. Fixes #417992.
2007-05-24 16:22:23 +00:00
Tim-Philipp Müller
57375cf664 gst-libs/gst/riff/riff-media.c: Don't add channels and rate fields to the template caps for audio/x-dts, as wavparse ...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Don't add channels and rate fields to the template caps for
audio/x-dts, as wavparse might not always be able to set them,
which would then lead to 'caps are not a real subset of the
template caps' warnings.
2007-05-24 15:16:59 +00:00
Jan Schmidt
bec7949e8e gst-libs/gst/riff/riff-media.c: Specify the full valid range for MP3 samplerates. Fixes a regression caused by extra ...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Specify the full valid range for MP3 samplerates. Fixes a regression
caused by extra header checks since the last release.
2007-05-22 11:40:31 +00:00
Wim Taymans
9b188adc27 Small cleanups.
Original commit message from CVS:
* ext/cdparanoia/gstcdparanoiasrc.c:
(gst_cd_paranoia_src_read_sector):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
Small cleanups.
* ext/theora/theoradec.c: (theora_dec_sink_event):
Fix typo.
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_set_gst_timestamp):
Add some FIXME
* gst/playback/gstdecodebin.c: (queue_underrun_cb):
And some debug info when a FIXME path is hit.
2007-05-21 10:25:44 +00:00
Wim Taymans
7ace85992a gst-libs/gst/rtp/gstbasertpaudiopayload.c: Some cleanups, remove minptime property as it is now in the parent class.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_class_init),
(gst_base_rtp_audio_payload_init),
(gst_base_rtp_audio_payload_finalize),
(gst_base_rtp_audio_payload_handle_frame_based_buffer),
(gst_base_rtp_audio_payload_handle_sample_based_buffer),
(gst_base_rtp_payload_audio_handle_event):
Some cleanups, remove minptime property as it is now in the parent
class.
Override parent class event function.
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_class_init), (gst_basertppayload_init),
(gst_basertppayload_event), (gst_basertppayload_set_property),
(gst_basertppayload_get_property):
* gst-libs/gst/rtp/gstbasertppayload.h:
Add min-ptime property.
Add handle-event vmethod. Fixes #415001.
2007-05-21 09:45:28 +00:00
Stefan Kost
e7c3ddf3fc gst-libs/gst/audio/gstbaseaudiosink.c
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c
(gst_base_audio_sink_change_state):
Fix typo in comment.
* gst/playback/gstdecodebin.c (gst_decode_bin_class_init,
free_dynamics, pad_probe, close_pad_link, try_to_link_1,
get_our_ghost_pad, remove_element_chain, queue_underrun_cb,
close_link):
* gst/playback/gstplaybin.c (gst_play_bin_set_property,
gen_audio_element, remove_sinks, gst_play_bin_send_event_to_sink):
Remove trailing whitespaces in comments.
* gst/volume/Makefile.am:
Fix tabs.
2007-05-18 15:23:43 +00:00
Marc-Andre Lureau
16b8bd4c49 gst-libs/gst/interfaces/mixer.h (mixer_type, option_changed, set_option, get_option, _gst_reserved):
Original commit message from CVS:
patch by: Marc-Andre Lureau <marcandre.lureau@gmail.com>
* gst-libs/gst/interfaces/mixer.h (mixer_type, option_changed,
set_option, get_option, _gst_reserved):
Revert reordering functions (keep ABI).
2007-05-18 15:10:08 +00:00
Marc-Andre Lureau
f2df2a6948 ext/alsa/gstalsamixer.c (main_context, GstAlsaMixerWatch, _GstAlsaMixerWatch, source, n_poll_fds, poll_fds, gst_alsa_...
Original commit message from CVS:
patch by: Marc-Andre Lureau <marcandre.lureau@gmail.com>
* ext/alsa/gstalsamixer.c (main_context, GstAlsaMixerWatch,
_GstAlsaMixerWatch, source, n_poll_fds, poll_fds,
gst_alsa_mixer_watch_funcs, gst_alsa_mixer_prepare,
gst_alsa_mixer_check, gst_alsa_mixer_dispatch,
gst_alsa_mixer_finalize, gst_alsa_mixer_find_master_mixer,
gst_alsa_mixer_handle_source_callback,
gst_alsa_mixer_handle_callback, gst_alsa_mixer_elem_handle_callback,
gst_alsa_mixer_ensure_track_list, gst_alsa_mixer_free,
gst_alsa_mixer_get_volume, gst_alsa_mixer_set_volume,
gst_alsa_mixer_set_mute, gst_alsa_mixer_set_record,
gst_alsa_mixer_get_option, gst_alsa_mixer_update_option,
gst_alsa_mixer_update_track, _gst_alsa_mixer_set_interface):
* ext/alsa/gstalsamixer.h (handle_source, interface, dir):
* ext/alsa/gstalsamixerelement.c (gst_alsa_mixer_element_details,
gst_alsa_mixer_element_interface_supported,
gst_alsa_mixer_element_finalize, gst_alsa_mixer_element_init,
gst_alsa_mixer_element_set_property,
gst_alsa_mixer_element_get_property,
gst_alsa_mixer_element_change_state):
* ext/alsa/gstalsamixertrack.c (gst_alsa_mixer_track_update):
* gst-libs/gst/interfaces/mixer.c (gst_mixer_volume_changed,
gst_mixer_option_changed):
* gst-libs/gst/interfaces/mixer.h (set_option, get_option,
volume_changed, option_changed, _gst_reserved):
Implement notification for alsamixer. Fixes #152864
2007-05-15 14:01:26 +00:00
Wim Taymans
01b6f0b353 gst-libs/gst/rtp/gstbasertpdepayload.c: Parse and use additional caps fields as described in updated application/x-rt...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_setcaps),
(gst_base_rtp_depayload_set_gst_timestamp):
Parse and use additional caps fields as described in updated
application/x-rtp caps spec.
2007-05-12 16:18:39 +00:00
Sébastien Moutte
c88306fe26 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Move variable declaration before the first instruction.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_handle_frame_based_buffer):
Move variable declaration before the first instruction.
* gst/videotestsrc/videotestsrc.c:
Define M_PI if it's not defined yet.
* win32/common/libgstrtp.def:
Add new exported functions.
2007-05-09 21:17:40 +00:00
Stefan Kost
64a9674bd2 gst/: gst/audiotestsrc/gstaudiotestsrc.c
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst/adder/gstadder.c:
* gst/audiotestsrc/gstaudiotestsrc.c
(gst_audio_test_src_create_white_noise):
* gst/videotestsrc/gstvideotestsrc.c:
* gst/volume/gstvolume.c (VOLUME_UNITY_INT16,
VOLUME_UNITY_INT16_BIT_SHIFT, VOLUME_MAX_DOUBLE,
volume_sink_template, volume_src_template, gst_volume_init,
volume_process_double, volume_process_int16,
volume_process_int16_clamp):
Doc fixes and formatting.
2007-05-04 13:10:07 +00:00
Tim-Philipp Müller
4f0e7a9ef9 gst-libs/gst/cdda/gstcddabasesrc.c: Fix it so that it (a) makes sense and (b) doesn't break everything cdda-related i...
Original commit message from CVS:
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_add_track):
Fix it so that it (a) makes sense and (b) doesn't break
everything cdda-related including the unit test.
2007-05-04 09:06:38 +00:00
Stefan Kost
57301524fb gst-libs/gst/cdda/gstcddabasesrc.c: Fix build when disabling asserts.
Original commit message from CVS:
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_add_track):
Fix build when disabling asserts.
2007-05-04 08:46:59 +00:00
Wim Taymans
7033e458ca gst-libs/gst/rtp/gstrtcpbuffer.c: Fix offset bug in generation RR packets.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_add_rb):
Fix offset bug in generation RR packets.
2007-04-29 14:38:05 +00:00
Wim Taymans
f23356bd8f gst-libs/gst/rtp/gstrtcpbuffer.*: Fix RB block parsing and writing.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_get_rb),
(gst_rtcp_packet_add_rb), (gst_rtcp_packet_sdes_add_item),
(gst_rtcp_packet_sdes_add_entry), (gst_rtcp_packet_bye_add_ssrc),
(gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset),
(gst_rtcp_packet_bye_set_reason):
* gst-libs/gst/rtp/gstrtcpbuffer.h:
Fix RB block parsing and writing.
Add support for constructing BYE packets.
2007-04-27 15:01:40 +00:00
Tim-Philipp Müller
9e873a3c83 When posting a warning message because samples were dropped, post something more intelligible than he default error m...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c: (_do_init),
(gst_base_audio_src_create):
* po/POTFILES.in:
When posting a warning message because samples were dropped, post
something more intelligible than he default error message for clock
errors which is just confusing in this context (#432984).
2007-04-25 08:54:34 +00:00
Wim Taymans
f5c743b069 gst-libs/gst/rtp/gstrtcpbuffer.*: Implement code to write SR, RR and SDES packets.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_buffer_new),
(gst_rtcp_buffer_end), (gst_rtcp_buffer_get_packet_count),
(read_packet_header), (gst_rtcp_packet_move_to_next),
(gst_rtcp_buffer_add_packet), (gst_rtcp_packet_sr_set_sender_info),
(gst_rtcp_packet_rr_set_ssrc), (gst_rtcp_packet_add_rb),
(gst_rtcp_packet_sdes_get_item_count),
(gst_rtcp_packet_sdes_first_item),
(gst_rtcp_packet_sdes_next_item), (gst_rtcp_packet_sdes_get_ssrc),
(gst_rtcp_packet_sdes_first_entry),
(gst_rtcp_packet_sdes_next_entry),
(gst_rtcp_packet_sdes_get_entry), (gst_rtcp_packet_sdes_add_item),
(gst_rtcp_packet_sdes_add_entry):
* gst-libs/gst/rtp/gstrtcpbuffer.h:
Implement code to write SR, RR and SDES packets.
2007-04-25 08:10:26 +00:00
Olivier Crete
e3ff444d30 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Chain up to parent class in dispose function; get rid of unnecessary 'dipo...
Original commit message from CVS:
Patch by: Olivier Crete  <tester at tester ca>
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_class_init),
(gst_base_rtp_audio_payload_init),
(gst_base_rtp_audio_payload_dispose):
Chain up to parent class in dispose function; get rid of
unnecessary 'diposed' flag in private structure (#415001).
2007-04-21 15:25:22 +00:00
Tim-Philipp Müller
71d77fbecc Some minor docs fixes and additions; also add missing 'Since' bits.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs.types:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_class_init):
* gst-libs/gst/rtp/gstbasertpdepayload.c:
* gst-libs/gst/rtp/gstbasertppayload.c:
Some minor docs fixes and additions; also add missing 'Since' bits.
2007-04-21 15:10:25 +00:00
Zeeshan Ali
80ebb9eb42 gst-libs/gst/rtp/gstbasertpaudiopayload.*: The recently-added gst_base_rtp_audio_payload_push() should take an object...
Original commit message from CVS:
Patch by: Zeeshan Ali  <zeenix gmail com>
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_handle_frame_based_buffer),
(gst_base_rtp_audio_payload_handle_sample_based_buffer),
(gst_base_rtp_audio_payload_push):
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
The recently-added gst_base_rtp_audio_payload_push() should take an
object of type GstBaseRTPAudioPayload as first argument (#431672).
2007-04-21 14:40:45 +00:00
Sebastian Dröge
9502a0e1b7 gst-libs/gst/riff/riff-media.c: Allow random depths between 1 and 32 instead of only multiplies of 8.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Allow random depths between 1 and 32 instead of only multiplies of 8.
2007-04-17 02:53:16 +00:00
Sebastian Dröge
cef23b669c gst-libs/gst/riff/riff-media.c: Set the maximum number of channels for PCM and float in the correct place to have it ...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Set the maximum number of channels for PCM and float in the correct
place to have it also used when creating the template caps.
2007-04-17 02:04:21 +00:00
Sebastian Dröge
3effb4d23e gst-libs/gst/riff/riff-media.c: Correctly support 4, 6 and 8 channels with normal PCM and float wav files.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Correctly support 4, 6 and 8 channels with normal PCM and float
wav files.
Fix the depth and signedness calculation in extensible wav files and
also handle 1, 2, 4, 6, 8 channels here when a file without channel
mask is found.
Add support for float, alaw and mulaw in extensible wav files.
This allows correct playback of all but 5 files from
http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/Samples.html
(gst_riff_create_audio_template_caps):
Add voxware and float formats to the template caps.
2007-04-17 01:56:07 +00:00
Vincent Torri
0138ad7e09 ext/pango/gstclockoverlay.c: Fix unused variable warning if HAVE_LOCALTIME_R is undefinied
Original commit message from CVS:
Patch by: Vincent Torri <vtorri at univ-evry dot fr>
* ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time):
Fix unused variable warning if HAVE_LOCALTIME_R is undefinied
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
* gst/audioresample/gstaudioresample.c: (audioresample_do_output):
Use the correct format strings for integer formats.
2007-04-16 22:20:03 +00:00
Stefan Kost
95ef089dc6 gst-libs/gst/riff/riff-media.c: More sanity checks for the header fields.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
More sanity checks for the header fields.
2007-04-13 06:17:45 +00:00
Tim-Philipp Müller
83ab98b0fc gst-libs/gst/tag/tags.c: Try encodings from all environment variables, not just those in the first environment variab...
Original commit message from CVS:
* gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
Try encodings from all environment variables, not just those in the
first environment variable that is set.
2007-04-12 16:36:36 +00:00
Tim-Philipp Müller
a208469078 API: add gst_tag_freeform_string_to_utf8() (#405072).
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/tag/tag.h:
* gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
API: add gst_tag_freeform_string_to_utf8() (#405072).
* gst-libs/gst/tag/gstid3tag.c: (gst_tag_extract_id3v1_string):
Use gst_tag_freeform_string_to_utf8() here.
2007-04-12 12:19:20 +00:00
Wim Taymans
b802dea831 gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to create invalid calibration parameters by making the internal time...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_skew_slaving):
Don't try to create invalid calibration parameters by making the
internal time go backwards, instead make external time go forward.
2007-04-05 15:44:40 +00:00
Sebastian Dröge
fac74a841b gst-libs/gst/riff/riff-media.c: Add audio/x-raw-float support, now that audioconvert support non-native endianness fl...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Add audio/x-raw-float support, now that audioconvert support
non-native endianness floats.
2007-03-30 17:05:23 +00:00
René Stadler
6ac8ff9ec3 with some minor changes
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler dot de>
with some minor changes
* gst-libs/gst/floatcast/floatcast.h:
Use more efficient float endianness conversion functions that don't
involve 2 function calls per value.
* gst/audioconvert/audioconvert.c: (audio_convert_get_func_index),
(check_default), (audio_convert_prepare_context):
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_parse_caps), (make_lossless_changes):
Support non-native endianness floats as input and output.
Fixes #339838.
* tests/check/elements/audioconvert.c: (verify_convert),
(GST_START_TEST):
Add unit tests for the non-native endianness float conversions.
2007-03-29 18:42:34 +00:00
Wim Taymans
76462ceb45 gst-libs/gst/rtp/gstbasertpdepayload.*: Add Private structure.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_base_init),
(gst_base_rtp_depayload_class_init), (gst_base_rtp_depayload_init),
(gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
(gst_base_rtp_depayload_set_gst_timestamp),
(gst_base_rtp_depayload_change_state),
(gst_base_rtp_depayload_set_property),
(gst_base_rtp_depayload_get_property):
* gst-libs/gst/rtp/gstbasertpdepayload.h:
Add Private structure.
Bring element code to 2007.
Parse clock-base caps param and use it when generating the
newsegment.
Reset variables before going to PAUSED.
Fix some docs.
2007-03-29 16:23:53 +00:00
Wim Taymans
0a39f494b5 Add RTCP docs.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_get_adapter):
Add RTCP docs.
Fix some more docs.
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/rtp/gstrtcpbuffer.c:
(gst_rtcp_buffer_new_take_data), (gst_rtcp_buffer_new_copy_data),
(gst_rtcp_buffer_validate_data), (gst_rtcp_buffer_validate),
(gst_rtcp_buffer_get_packet_count), (read_packet_header),
(gst_rtcp_buffer_get_first_packet), (gst_rtcp_packet_move_to_next),
(gst_rtcp_buffer_add_packet), (gst_rtcp_packet_remove),
(gst_rtcp_packet_get_padding), (gst_rtcp_packet_get_type),
(gst_rtcp_packet_get_count), (gst_rtcp_packet_get_length),
(gst_rtcp_packet_sr_get_sender_info),
(gst_rtcp_packet_sr_set_sender_info),
(gst_rtcp_packet_rr_get_ssrc), (gst_rtcp_packet_rr_set_ssrc),
(gst_rtcp_packet_get_rb_count), (gst_rtcp_packet_get_rb),
(gst_rtcp_packet_add_rb), (gst_rtcp_packet_set_rb),
(gst_rtcp_packet_sdes_get_chunk_count),
(gst_rtcp_packet_sdes_first_chunk),
(gst_rtcp_packet_sdes_next_chunk), (gst_rtcp_packet_sdes_get_ssrc),
(gst_rtcp_packet_sdes_first_item),
(gst_rtcp_packet_sdes_next_item), (gst_rtcp_packet_sdes_get_item),
(gst_rtcp_packet_bye_get_ssrc_count),
(gst_rtcp_packet_bye_get_nth_ssrc), (gst_rtcp_packet_bye_add_ssrc),
(gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset),
(gst_rtcp_packet_bye_get_reason_len),
(gst_rtcp_packet_bye_get_reason), (gst_rtcp_packet_bye_set_reason):
* gst-libs/gst/rtp/gstrtcpbuffer.h:
Add new helper object for parsing and creating RTCP messages.
2007-03-29 16:20:31 +00:00
Sebastian Dröge
dfdd873f6a gst-libs/gst/riff/riff-media.c: PCM samples with width=8 must be always unsigned, no matter what depth they have.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
PCM samples with width=8 must be always unsigned, no matter what
depth they have.
2007-03-29 12:07:02 +00:00
Wim Taymans
d4015266aa gst-libs/gst/riff/riff-ids.h: Add some more RIFF formats.
Original commit message from CVS:
* gst-libs/gst/riff/riff-ids.h:
Add some more RIFF formats.
2007-03-29 10:19:45 +00:00
Wim Taymans
804e7d1759 gst-libs/gst/rtp/gstrtpbuffer.*: Fix fixed payload names and docs.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtpbuffer.c:
(gst_rtp_buffer_default_clock_rate):
* gst-libs/gst/rtp/gstrtpbuffer.h:
Fix fixed payload names and docs.
Added method to get the default clock rates of fixed payload types.
API: GstRTPBuffer::gst_rtp_buffer_default_clock_rate()
2007-03-29 10:17:52 +00:00
Wim Taymans
450030ebaf gst-libs/gst/audio/gstbaseaudiosink.*: Store private stuff in GstBaseAudioSinkPrivate.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c: (slave_method_get_type),
(gst_base_audio_sink_class_init), (gst_base_audio_sink_init),
(gst_base_audio_sink_query), (gst_base_audio_sink_get_time),
(gst_base_audio_sink_set_property),
(gst_base_audio_sink_get_property), (gst_base_audio_sink_event),
(clock_convert_external), (gst_base_audio_sink_resample_slaving),
(gst_base_audio_sink_skew_slaving),
(gst_base_audio_sink_handle_slaving), (gst_base_audio_sink_render),
(gst_base_audio_sink_async_play):
* gst-libs/gst/audio/gstbaseaudiosink.h:
Store private stuff in GstBaseAudioSinkPrivate.
Add configurable clock slaving modes property.
API:: GstBaseAudioSink::slave-method property
Some more latency reporting tweaks.
Added skew based clock slaving correction and make it the default until
the resampling method is more robust.
2007-03-28 14:50:47 +00:00
Tim-Philipp Müller
726f2c1732 Make sure we parse floating-point numbers in vorbis comments correctly with either '.' or ',' as separator, no matter...
Original commit message from CVS:
* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
* tests/check/libs/tag.c: (GST_START_TEST):
Make sure we parse floating-point numbers in vorbis comments
correctly with either '.' or ',' as separator, no matter what
the current locale is. Add unit test for this too.
2007-03-27 10:17:16 +00:00
René Stadler
01a1e4bc81 gst-libs/gst/tag/gstvorbistag.c: When writing out floating-point numbers to vorbis comment tags, always use the same ...
Original commit message from CVS:
Patch by: René Stadler  <mail at renestadler de>
* gst-libs/gst/tag/gstvorbistag.c: (gst_tag_to_vorbis_comments):
When writing out floating-point numbers to vorbis comment tags, always
use the same character as separator no matter what the current locale is
(fixes #423051).
* tests/check/libs/tag.c: (GST_START_TEST):
Add unit tests for replaygain tags in vorbis comments (closes #423055).
2007-03-26 22:38:19 +00:00
Jan Schmidt
77683331e1 gst-libs/gst/rtp/Makefile.am: The base audio payloader uses GstAdapter - we need GST_BASE_LIBS.
Original commit message from CVS:
* gst-libs/gst/rtp/Makefile.am:
The base audio payloader uses GstAdapter - we need GST_BASE_LIBS.
2007-03-26 11:44:07 +00:00
Thomas Vander Stichele
1e467ec211 gst-libs/gst/app/Makefile.am: Use GST_ALL_LDFLAGS, which actually exists, but maybe David can confirm that was what h...
Original commit message from CVS:
* gst-libs/gst/app/Makefile.am:
Use GST_ALL_LDFLAGS, which actually exists, but maybe David
can confirm that was what he wanted.
2007-03-22 14:37:08 +00:00
Philippe Kalaf
b6d7f65463 gst-libs/gst/rtp/gstbasertpaudiopayload.*: olivier.crete@collabora.co.uk.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
Add min-ptime property to RTP base audio payloader. Patch by
olivier.crete@collabora.co.uk.
Fixes #415001

Indentation/whitespace/documentation fixes.
2007-03-14 21:11:18 +00:00
David Schleef
6cf863e33c Add appsrc/appsink example.
Original commit message from CVS:
* configure.ac:
* examples/Makefile.am:
* examples/app/Makefile.am:
* examples/app/appsrc_ex.c:
Add appsrc/appsink example.
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/app/gstapp.c:
* gst-libs/gst/app/gstappsink.c:
* gst-libs/gst/app/gstappsink.h:
* gst/app/gstapp.c:
Add appsink.
2007-03-11 00:48:26 +00:00
Sébastien Moutte
1596dd263c gst-libs/gst/audio/gstbaseaudiosink.c: Use gst_guint64_to_gdouble for conversion.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:(gst_base_audio_sink_render):
Use gst_guint64_to_gdouble for conversion.
* win32/MANIFEST:
Add new files to the win32 MANIFEST.
* win32/common/libgstaudio.def:
* win32/common/libgstpbutils.def:
Add new exported functions.
* win32/vs6/gst_plugins_base.dsw:
* win32/vs6/libgstdecodebin.dsp:
* win32/vs6/libgstplaybin.dsp:
Change the link to libgstpbutils.lib.
* win32/vs6/libgstdecodebin2.dsp:
Add a new project for decodebin2.
* win32/vs6/libgstpbutils.dsp:
Add a new project for pbutils.
2007-03-10 15:59:33 +00:00
Tim-Philipp Müller
4462906be4 gst-libs/gst/tag/gstvorbistag.c: Also accept partial dates with only year and month, like 1999-12-00 (fixes #410396 e...
Original commit message from CVS:
* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
Also accept partial dates with only year and month,
like 1999-12-00 (fixes #410396 even more).
* tests/check/libs/tag.c: (GST_START_TEST):
Add unit test for the above.
2007-03-10 12:18:58 +00:00
Wim Taymans
5676bdaf81 gst-libs/gst/tag/gstid3tag.c: Add read support for GST_TAG_MUSICBRAINZ_SORTNAME (TSOP) tag.
Original commit message from CVS:
* gst-libs/gst/tag/gstid3tag.c:
Add read support for GST_TAG_MUSICBRAINZ_SORTNAME (TSOP) tag.
Fixes #414496.
2007-03-09 16:51:13 +00:00
Wim Taymans
e9be846621 Use new metadata copy function.
Original commit message from CVS:
* ext/pango/gsttextrender.c: (gst_text_render_chain):
* ext/vorbis/vorbistag.c: (gst_vorbis_tag_parse_packet):
* gst-libs/gst/netbuffer/gstnetbuffer.c: (gst_netbuffer_copy):
Use new metadata copy function.
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_transform):
* gst/videoscale/gstvideoscale.c: (gst_video_scale_transform):
Basetransform copied the metadata for us.
2007-03-09 16:38:06 +00:00
Tim-Philipp Müller
4aa8b0ca21 gst-libs/gst/interfaces/mixertrack.c: API: add "untranslated-label" property which should be set by implementations a...
Original commit message from CVS:
* gst-libs/gst/interfaces/mixertrack.c:
(gst_mixer_track_class_init), (gst_mixer_track_get_property),
(gst_mixer_track_set_property):
API: add "untranslated-label" property which should be set by
implementations at construct time (#414645).
* ext/alsa/gstalsamixeroptions.c: (gst_alsa_mixer_options_new):
* ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
Set "untranslated-label" when constructing mixer track objects.
* tests/check/elements/alsa.c: (GST_START_TEST), (alsa_suite):
Unit test to check the above.
2007-03-07 18:50:10 +00:00
Wim Taymans
a2a8b1b8ce gst-libs/gst/audio/gstbaseaudiosink.c: Fix regression that made GStreamer skip the first samples of audio.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_async_play):
Fix regression that made GStreamer skip the first samples of audio.
Fixes #414684.
2007-03-06 12:10:08 +00:00
Tim-Philipp Müller
5d14dbbcda gst-libs/gst/pbutils/Makefile.am: Change directory to install headers in from gst/utils to gst/pbutils as well.
Original commit message from CVS:
* gst-libs/gst/pbutils/Makefile.am:
Change directory to install headers in from gst/utils to gst/pbutils
as well.
2007-03-05 09:27:55 +00:00
Thomas Vander Stichele
f6bd20e5e3 rename utils to pbutils
Original commit message from CVS:
* configure.ac:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/Makefile.am:
* gst-libs/gst/interfaces/mixer.c:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/pbutils/descriptions.c:
(gst_pb_utils_get_source_description),
(gst_pb_utils_get_sink_description),
(gst_pb_utils_get_decoder_description),
(gst_pb_utils_get_encoder_description),
(gst_pb_utils_get_element_description),
(gst_pb_utils_add_codec_description_to_tag_list),
(gst_pb_utils_get_codec_description), (gst_pb_utils_list_all):
* gst-libs/gst/pbutils/descriptions.h:
* gst-libs/gst/pbutils/install-plugins.c:
* gst-libs/gst/pbutils/install-plugins.h:
* gst-libs/gst/pbutils/missing-plugins.c:
(gst_missing_uri_source_message_new),
(gst_missing_uri_sink_message_new),
(gst_missing_element_message_new),
(gst_missing_decoder_message_new),
(gst_missing_encoder_message_new),
(gst_missing_plugin_message_get_description):
* gst-libs/gst/pbutils/missing-plugins.h:
* gst-libs/gst/pbutils/pbutils.c: (gst_pb_utils_init):
* gst-libs/gst/pbutils/pbutils.h:
* gst-libs/gst/utils/Makefile.am:
* gst-libs/gst/utils/base-utils.c:
* gst-libs/gst/utils/base-utils.h:
* gst-libs/gst/utils/descriptions.c:
* gst-libs/gst/utils/descriptions.h:
* gst-libs/gst/utils/install-plugins.c:
* gst-libs/gst/utils/install-plugins.h:
* gst-libs/gst/utils/missing-plugins.c:
* gst-libs/gst/utils/missing-plugins.h:
* gst-plugins-base.spec.in:
* gst/playback/Makefile.am:
* gst/playback/gstdecodebin.c:
* gst/playback/gstdecodebin2.c:
* gst/playback/gstplaybasebin.c: (setup_subtitle),
(gen_source_element):
* gst/playback/gstplaybin.c: (plugin_init):
* tests/check/Makefile.am:
* tests/check/libs/pbutils.c: (GST_START_TEST),
(test_pb_utils_install_plugins_do_callout), (libgstpbutils_suite):
* tests/check/libs/utils.c:
rename utils to pbutils
2007-03-04 23:39:51 +00:00
David Schleef
64d706f402 gst-libs/gst/app/Makefile.am: Install the headers.
Original commit message from CVS:
* gst-libs/gst/app/Makefile.am:
Install the headers.
2007-03-03 10:23:03 +00:00
David Schleef
b11893bb27 gst-libs/gst/app/: Add GstAppBuffer that includes a callback and closure for proper handling of data chunks.
Original commit message from CVS:
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/app/gstappbuffer.c:
* gst-libs/gst/app/gstappbuffer.h:
* gst-libs/gst/app/gstappsrc.c:
Add GstAppBuffer that includes a callback and closure for
proper handling of data chunks.
2007-03-03 10:10:30 +00:00
David Schleef
e8afc8b284 gst-libs/gst/app/gstappsrc.*: Hacking to address issues in 413418.
Original commit message from CVS:
* gst-libs/gst/app/gstappsrc.c:
* gst-libs/gst/app/gstappsrc.h:
Hacking to address issues in 413418.
2007-03-03 09:06:06 +00:00
David Schleef
2164be520c Move the app library to gst-libs/gst/app (duh!)
Original commit message from CVS:
* Makefile.am:
* configure.ac:
* ext/Makefile.am:
* gst-libs/gst/Makefile.am:
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/app/gstapp.c:
* gst-libs/gst/app/gstappsrc.c:
* gst-libs/gst/app/gstappsrc.h:
* gst/app/Makefile.am:
* gst/app/gstapp.c:
* gst/app/gstappsrc.c:
* gst/app/gstappsrc.h:
Move the app library to gst-libs/gst/app (duh!)
2007-03-03 08:16:57 +00:00
Wim Taymans
5ee0a694a6 gst-libs/gst/audio/gstbaseaudiosink.c: base time is irrelevant here.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_async_play):
base time is irrelevant here.
2007-03-01 17:29:55 +00:00
Wim Taymans
85c7eeecc3 gst-libs/gst/audio/: Improve debugging.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
* gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
Improve debugging.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_query), (gst_base_audio_sink_event),
(gst_base_audio_sink_render), (gst_base_audio_sink_async_play):
Improve latency and clock slaving calculations.
Improve slave clock calibration.
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_commit_full):
When we are asked to render N sample to 0 bytes, return N.
2007-03-01 17:01:43 +00:00
Wim Taymans
3c94c06c5a gst-libs/gst/audio/gstaudioclock.c: Fix clock name.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
(gst_audio_clock_new):
Fix clock name.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_init), (gst_base_audio_sink_query):
* gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
(gst_base_audio_src_query), (gst_base_audio_src_get_offset),
(gst_base_audio_src_create):
Improve latency query code.
Use proper clock names.
2007-02-28 15:02:25 +00:00
René Stadler
88e94fc278 gst-libs/gst/tag/gstvorbistag.c: Parse date strings in vorbis comments that have an invalid (zero) month or day (#410...
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler de>
* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
Parse date strings in vorbis comments that have an invalid (zero)
month or day (#410396).
* tests/check/libs/tag.c: (GST_START_TEST):
Test case for the above.
2007-02-25 23:51:03 +00:00
Tim-Philipp Müller
e8e648a76d Change GStreamer marker prefix in detail string from 'gstreamer.net' to just 'gstreamer'. Document the caps string co...
Original commit message from CVS:
* gst-libs/gst/utils/install-plugins.c:
* gst-libs/gst/utils/missing-plugins.c:
* tests/check/libs/utils.c: (missing_msg_check_getters):
Change GStreamer marker prefix in detail string from 'gstreamer.net'
to just 'gstreamer'. Document the caps string component of the
decoder/encoder detail a bit better, since not everyone will be
familiar with the GStreamer media type/caps system (but they better
enjoy nested itemized lists).
2007-02-23 13:10:50 +00:00
Tim-Philipp Müller
011471dbbb gst-libs/gst/netbuffer/gstnetbuffer.c: Fix copying of GstNetBuffer (would crash before, or at least lead to invalid m...
Original commit message from CVS:
* gst-libs/gst/netbuffer/gstnetbuffer.c:
(notgst_buffer_copy_fields_in_place), (gst_netbuffer_copy):
Fix copying of GstNetBuffer (would crash before, or at least lead to
invalid memory access, #410772), for now by copying the GstBuffer copy
code from the core over here so we can copy the GstBuffer fields on a
provided buffer instance (of type GstNetBuffer in this case). Would be
better to fix this with some support by the core though (and in the long
run change the broken GstBuffer/GstMiniObject copy semantics, #393099).
* tests/check/Makefile.am:
Enable unit test for GstNetBuffer.
2007-02-22 12:57:47 +00:00
Andy Wingo
d9b6796d91 gst-libs/gst/audio/gstbaseaudiosink.c
Original commit message from CVS:
2007-02-22  Andy Wingo  <wingo@pobox.com>

* gst-libs/gst/audio/gstbaseaudiosink.c
(gst_base_audio_sink_init): Disable pull-mode activation until we
figure out how to make audio sinks go to PLAYING.
2007-02-22 11:04:10 +00:00
Tim-Philipp Müller
b99629643c gst-libs/gst/utils/: Some more docs (and descriptions for two subtitle formats).
Original commit message from CVS:
* gst-libs/gst/utils/base-utils.c:
* gst-libs/gst/utils/descriptions.c:
* gst-libs/gst/utils/install-plugins.c:
* gst-libs/gst/utils/missing-plugins.c:
Some more docs (and descriptions for two subtitle formats).
2007-02-18 21:02:36 +00:00
Tim-Philipp Müller
2f45e10c73 gst-libs/gst/audio/audio.c: Fix documentation.
Original commit message from CVS:
* gst-libs/gst/audio/audio.c:
Fix documentation.
2007-02-16 10:19:45 +00:00
Stefan Kost
b2f9c0f289 More docs coverage and some ChangeLog surgery (add missing names)
Original commit message from CVS:
* ext/cdparanoia/gstcdparanoiasrc.h:
* ext/ogg/gstoggdemux.h:
* gst-libs/gst/audio/audio.c: (gst_audio_frame_byte_size),
(gst_audio_frame_length), (gst_audio_duration_from_pad_buffer),
(gst_audio_is_buffer_framed), (gst_audio_structure_set_int):
* gst-libs/gst/audio/audio.h:
* gst-libs/gst/audio/gstaudiofilter.h:
* gst-libs/gst/interfaces/videoorientation.h:
* gst/adder/gstadder.h:
More docs coverage and some ChangeLog surgery (add missing names)
2007-02-15 15:17:23 +00:00
Wim Taymans
a43d0f57eb gst-libs/gst/audio/gstbaseaudiosink.c: Answer latency query.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init), (gst_base_audio_sink_query),
(gst_base_audio_sink_render), (gst_base_audio_sink_callback),
(gst_base_audio_sink_async_play),
(gst_base_audio_sink_change_state):
Answer latency query.
Use configured latency when syncing.
Fix clock slaving.
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init), (gst_base_audio_src_dispose),
(gst_base_audio_src_query), (gst_base_audio_src_change_state):
Fix possible memleak.
Implement latency query.
Small cleanups.
2007-02-15 12:06:25 +00:00
Stefan Kost
7ee1b714f0 Add crossreferences to glib/gobject/gstream docs.
Original commit message from CVS:
* configure.ac:
* docs/libs/Makefile.am:
* docs/plugins/Makefile.am:
Add crossreferences to glib/gobject/gstream docs.
* gst-libs/gst/audio/audio.h:
Source formatting.
* gst/audiotestsrc/gstaudiotestsrc.c: (plugin_init):
Add own debug category.
2007-02-12 20:42:23 +00:00
René Stadler
14f2d7efdb gst-libs/gst/tag/gstvorbistag.c: Add vorbis/FLAC-tag mapping for new GST_TAG_REFERENCE_LEVEL (#403597).
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler de>
* gst-libs/gst/tag/gstvorbistag.c:
Add vorbis/FLAC-tag mapping for new GST_TAG_REFERENCE_LEVEL
(#403597).
2007-02-12 11:01:04 +00:00
Sébastien Moutte
9caee48ed4 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Use gst_gdouble_to_guint64 for conversions.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:(gst_base_rtp_audio_payload_handle_frame_based_buffer):
Use gst_gdouble_to_guint64 for conversions.
* win32/common/config.h.in:
Add a define for GST_INSTALL_PLUGINS_HELPER
* win32/common/libgstaudio.def:
* win32/common/libgstcdda.def:
* win32/common/libgstnetbuffer.def:
* win32/common/libgstrtp.def:
* win32/common/libgutils.def:
Add new exported functions.
* win32/vs6/gst_plugins_base.dsw:
* win32/vs6/libgstdecodebin.dsp:
* win32/vs6/libgstnetbuffer.dsp:
* win32/vs6/libgstplaybin.dsp:
* win32/vs6/libgstrtp.dsp:
* win32/vs6/libgstvorbis.dsp:
* win32/vs6/libgstcdda.dsp:
* win32/vs6/libgstgdp.dsp:
* win32/vs6/libgstutils.dsp:
Update and add new project files.
2007-02-10 19:27:48 +00:00
Tim-Philipp Müller
5b499dec66 gst-libs/gst/audio/gstaudiofilter.c: Clear our formats structure and free the caps contained in it when shutting down.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiofilter.c:
(gst_audio_filter_class_init), (gst_audio_filter_change_state):
Clear our formats structure and free the caps contained in it when
shutting down.
2007-02-06 09:42:05 +00:00
Andy Wingo
451ff2f992 gst-libs/gst/audio/gstbaseaudiosink.c
Original commit message from CVS:
2007-02-05  Andy Wingo  <wingo@pobox.com>

* gst-libs/gst/audio/gstbaseaudiosink.c
(gst_base_audio_sink_callback): Update basesink->offset so that we
pull monotonically increasing offsets instead of, um, seeking back
to 0 each time. Fixes alsasrc ! alsasink!
2007-02-05 18:39:51 +00:00
Tim-Philipp Müller
2594880e87 gst-libs/gst/audio/: Oops, forgot to commit fixed-up example.
Original commit message from CVS:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
(gst_audio_filter_template_base_init),
(gst_audio_filter_template_class_init),
(gst_audio_filter_template_init),
(gst_audio_filter_template_set_property),
(gst_audio_filter_template_get_property),
(gst_audio_filter_template_setup),
(gst_audio_filter_template_filter),
(gst_audio_filter_template_filter_inplace), (plugin_init):
Oops, forgot to commit fixed-up example.
2007-02-03 23:28:45 +00:00
Tim-Philipp Müller
b63fff63d4 Port GstAudioFilter to 0.10. This change technically breaks but seems justifiable on the grounds that the base class ...
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_get_type),
(gst_audio_filter_class_init), (gst_audio_filter_init),
(gst_audio_filter_set_caps),
(gst_audio_filter_class_add_pad_templates):
* gst-libs/gst/audio/gstaudiofilter.h:
Port GstAudioFilter to 0.10. This change technically breaks
API and ABI (and thus also every library developer's heart),
but seems justifiable on the grounds that the base class was
completely unusable before (ie. would crash immediately when
actually used). Fixes #403963 (and eventually also #403572).
Also document all of this a bit.
2007-02-03 20:19:35 +00:00
Tim-Philipp Müller
7d78598f24 Lowering log level to see why things fail on the p5 build bot; fix some typos in unit test messages.
Original commit message from CVS:
* gst-libs/gst/utils/install-plugins.c:
(gst_install_plugins_spawn_child):
* tests/check/libs/utils.c:
(test_base_utils_install_plugins_do_callout):
Lowering log level to see why things fail on the p5 build bot;
fix some typos in unit test messages.
2007-02-03 14:26:54 +00:00
Tim-Philipp Müller
17a02da2fd gst-libs/gst/utils/: API: add API for applications to initiate installation of missing plugins, ie. gst_install_plugi...
Original commit message from CVS:
* gst-libs/gst/utils/Makefile.am:
* gst-libs/gst/utils/base-utils.h:
* gst-libs/gst/utils/install-plugins.c:
(gst_install_plugins_context_set_xid),
(gst_install_plugins_context_new),
(gst_install_plugins_context_free),
(gst_install_plugins_get_helper),
(gst_install_plugins_spawn_child),
(gst_install_plugins_return_from_status),
(gst_install_plugins_installer_exited),
(gst_install_plugins_async), (gst_install_plugins_sync),
(gst_install_plugins_return_get_name),
(gst_install_plugins_installation_in_progress):
* gst-libs/gst/utils/install-plugins.h:
API: add API for applications to initiate installation of missing
plugins, ie. gst_install_plugins_async() primarily.
Based on libgimme-codec by Ryan Lortie.
* configure.ac:
Add --with-install-plugins-helper configure option so distros can specify
the path of the helper script or program to call when plugin installation
is requested (distros: please do any argument munging in this helper
script instead of patching GStreamer to pass arguments differently
to another program directly).
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
Build and document new API.
* tests/check/libs/utils.c: (result_cb),
(test_base_utils_install_plugins_do_callout), (GST_START_TEST),
(libgstbaseutils_suite):
Some simple checks for the new API.
2007-02-02 20:42:08 +00:00
Wim Taymans
81e92118da gst-libs/gst/rtp/gstrtpbuffer.h: Add some more fixed payloads.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtpbuffer.h:
Add some more fixed payloads.
2007-01-24 12:10:56 +00:00
Tim-Philipp Müller
58e6e134cb gst-libs/gst/riff/riff-media.c: On second thought, use "depth" field rather than "bpp" field.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
On second thought, use "depth" field rather than "bpp" field.
2007-01-22 10:27:26 +00:00
Tim-Philipp Müller
439b3193bd gst-libs/gst/riff/riff-media.c: Camtasia caps apparently need a bpp field (#398875).
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
Camtasia caps apparently need a bpp field (#398875).
2007-01-22 09:23:01 +00:00
Tim-Philipp Müller
0eac623115 gst/: Fix potentially unaligned access (#397207).
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
* gst-libs/gst/tag/gstvorbistag.c:
(gst_tag_list_to_vorbiscomment_buffer):
* gst/typefind/gsttypefindfunctions.c: (vorbis_type_find):
Fix potentially unaligned access (#397207).
2007-01-16 19:37:55 +00:00
Stefan Kost
9f6e8af294 gst-libs/gst/tag/: Use new beats-per-minute tag from core.
Original commit message from CVS:
* gst-libs/gst/tag/gstid3tag.c:
* gst-libs/gst/tag/gstvorbistag.c:
Use new beats-per-minute tag from core.
2007-01-15 13:58:58 +00:00
Andy Wingo
d853b23819 gst-libs/gst/audio/gstbaseaudiosink.c
Original commit message from CVS:
2007-01-12  Andy Wingo  <wingo@pobox.com>

* gst-libs/gst/audio/gstbaseaudiosink.c
(gst_base_audio_sink_fixate): Implement, stolen from baseaudiosrc.
(gst_base_audio_sink_activate_pull): Remove the handwavey nego
stuff, as the base class handles this now. Actually tell the ring
buffer to start.
(gst_base_audio_sink_callback): Cast the ring buffer correctly.
How did this work before? Maybe I'm not as awesome a programmer as
I think.

* gst-libs/gst/audio/gstbaseaudiosrc.c
(gst_base_audio_src_fixate): Rework as a basesrc vmethod instead
of a pad function.
2007-01-12 21:19:35 +00:00
Tim-Philipp Müller
b93a9176db gst-libs/gst/utils/missing-plugins.c: Remove more fields so that the application can better blacklist formats that ha...
Original commit message from CVS:
* gst-libs/gst/utils/missing-plugins.c: (copy_and_clean_caps):
Remove more fields so that the application can better blacklist
formats that have been tried before.
2007-01-12 18:08:23 +00:00
Tim-Philipp Müller
ddf40c2406 gst-libs/gst/audio/mixerutils.h: Add G_BEGIN_DECLS and G_END_DECLS guards so these helpers can be used when compiling...
Original commit message from CVS:
* gst-libs/gst/audio/mixerutils.h:
Add G_BEGIN_DECLS and G_END_DECLS guards so these helpers can be
used when compiling with c++ compilers as well.
2007-01-12 12:47:29 +00:00
Tim-Philipp Müller
1450f0fb18 API: add new libgstbaseutils library with functions
Original commit message from CVS:
* configure.ac:
* gst-libs/gst/Makefile.am:
* gst-libs/gst/utils/Makefile.am:
* gst-libs/gst/utils/base-utils.c: (gst_base_utils_init):
* gst-libs/gst/utils/base-utils.h:
* gst-libs/gst/utils/descriptions.c: (format_info_get_desc),
(find_format_info), (caps_are_rtp_caps),
(gst_base_utils_get_source_description),
(gst_base_utils_get_sink_description),
(gst_base_utils_get_decoder_description),
(gst_base_utils_get_encoder_description),
(gst_base_utils_get_element_description),
(gst_base_utils_add_codec_description_to_tag_list),
(gst_base_utils_get_codec_description), (gst_base_utils_list_all):
* gst-libs/gst/utils/descriptions.h:
* gst-libs/gst/utils/missing-plugins.c:
(missing_structure_get_type), (copy_and_clean_caps),
(gst_missing_uri_source_message_new),
(gst_missing_uri_sink_message_new),
(gst_missing_element_message_new),
(gst_missing_decoder_message_new),
(gst_missing_encoder_message_new),
(missing_structure_get_string_detail),
(missing_structure_get_caps_detail),
(gst_missing_plugin_message_get_installer_detail),
(gst_missing_plugin_message_get_description),
(gst_is_missing_plugin_message):
* gst-libs/gst/utils/missing-plugins.h:
API: add new libgstbaseutils library with functions
- to create and parse missing-plugins messages
- that provide (translated) descriptions for caps/decoders/sources/etc.
Closes #392393.
* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
* pkgconfig/gstreamer-plugins-base.pc.in:
Add new lib.
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
Generate docs for new lib and API.
* tests/check/Makefile.am:
* tests/check/libs/.cvsignore:
* tests/check/libs/utils.c: (missing_msg_check_getters),
(GST_START_TEST), (libgstbaseutils_suite):
Add some basic unit tests.
2007-01-09 14:20:08 +00:00
Wim Taymans
62ef7da73b Small documentation updates/fixes
Original commit message from CVS:
* ext/theora/theoradec.c:
* ext/vorbis/vorbisdec.c:
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_commit_full):
* gst-libs/gst/audio/gstringbuffer.h:
* gst-libs/gst/rtp/gstrtpbuffer.c:
* gst-libs/gst/tag/gstvorbistag.c:
Small documentation updates/fixes
2007-01-09 11:15:57 +00:00
Andy Wingo
85aee8e273 gst-libs/gst/audio/gstbaseaudiosink.c (gst_base_audio_sink_class_init)
Original commit message from CVS:
2007-01-06  Andy Wingo  <wingo@pobox.com>

* gst-libs/gst/audio/gstbaseaudiosink.c
(gst_base_audio_sink_class_init)
(gst_base_audio_sink_init):
(gst_base_audio_sink_activate_pull): Add an activate_pull function
to baseaudiosink, and tell basesink that we can work in pull mode.
This way the ring buffer thread drives the pipeline directly, if
pull mode is possible. There is some lingering nastiness regarding
capsnego, however.
(gst_base_audio_sink_callback): Implement the callback to pull
data. This interface is a bit light, though -- it should get a
GstFlowReturn return value at least.
2007-01-06 17:28:40 +00:00
Thomas Vander Stichele
95ada43982 configure.ac: split out GST_CFLAGS into GST_PLUGINS_BASE_CFLAGS and GST_CFLAGS so that GST_BASE_CFLAGS can go inbetwe...
Original commit message from CVS:
* configure.ac:
split out GST_CFLAGS into GST_PLUGINS_BASE_CFLAGS and GST_CFLAGS
so that GST_BASE_CFLAGS can go inbetween them, making sure
we use uninstalled gst-libs headers
* docs/libs/Makefile.am:
* ext/alsa/Makefile.am:
* ext/cdparanoia/Makefile.am:
* ext/gnomevfs/Makefile.am:
* ext/libvisual/Makefile.am:
* ext/ogg/Makefile.am:
* ext/theora/Makefile.am:
* ext/vorbis/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/cdda/Makefile.am:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/tag/Makefile.am:
* gst/adder/Makefile.am:
* gst/audioconvert/Makefile.am:
* gst/audiorate/Makefile.am:
* gst/audioresample/Makefile.am:
* gst/playback/Makefile.am:
* gst/tcp/Makefile.am:
* gst/videoscale/Makefile.am:
* gst/volume/Makefile.am:
* sys/ximage/Makefile.am:
* sys/xvimage/Makefile.am:
* tests/icles/Makefile.am:
adapt
2007-01-04 12:49:48 +00:00
Julien Moutte
163ec9ecf9 Add a method to the XOverlay interface to allow disabling of event handling in x[v]imagesink elements. This will let ...
Original commit message from CVS:
2007-01-04  Julien MOUTTE  <julien@moutte.net>

* gst-libs/gst/interfaces/xoverlay.c:
(gst_x_overlay_handle_events):
* gst-libs/gst/interfaces/xoverlay.h:
* sys/ximage/ximagesink.c: (gst_ximagesink_xwindow_new),
(gst_ximagesink_set_xwindow_id),
(gst_ximagesink_set_event_handling),
(gst_ximagesink_xoverlay_init), (gst_ximagesink_set_property),
(gst_ximagesink_get_property), (gst_ximagesink_init),
(gst_ximagesink_class_init):
* sys/ximage/ximagesink.h:
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xwindow_new),
(gst_xvimagesink_set_xwindow_id),
(gst_xvimagesink_set_event_handling),
(gst_xvimagesink_xoverlay_init), (gst_xvimagesink_set_property),
(gst_xvimagesink_get_property), (gst_xvimagesink_init),
(gst_xvimagesink_class_init):
* sys/xvimage/xvimagesink.h:
* tests/icles/stress-xoverlay.c: (toggle_events),
(create_window):
Add a method to the XOverlay interface to allow disabling of
event handling in x[v]imagesink elements. This will let X events
propagate to parent windows which can be usefull in some cases.
Be carefull that the application is then responsible of pushing
navigation events and expose events to the video sink.
Fixes: #387138.
2007-01-04 11:30:53 +00:00
Tim-Philipp Müller
5c14969645 Add vorbistag <=> GStreamer tag mapping for GST_TAG_LOCATION (fixes #392070).
Original commit message from CVS:
* gst-libs/gst/tag/gstvorbistag.c:
* tests/check/libs/tag.c: (GST_START_TEST):
Add vorbistag <=> GStreamer tag mapping for GST_TAG_LOCATION
(fixes #392070).
2007-01-03 15:45:06 +00:00
Tim-Philipp Müller
20862a8523 docs/libs/gst-plugins-base-libs-sections.txt: Small docs fixes/updates.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
Small docs fixes/updates.
* gst-libs/gst/video/gstvideosink.h:
Remove nonfunctional GST_VIDEO_SINK_CLOCK macro which is a leftover
from the 0.9 days (GST_BASE_SINK_CLOCK, which it points to, was
removed from the base sink API between 0.9.6 and 0.9.7).
API: add GST_VIDEO_SINK_CAST and use it for the height/width
accessor macros, so we don't do a runtime GObject type check every
time we use them.
2006-12-15 10:52:23 +00:00
Jens Granseuer
595217e840 Declare variables at the beginning of a block. Fixes #383195.
Original commit message from CVS:
Patch by: Jens Granseuer <jensgr at gmx net>
* gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_handle_frame_based_buffer),
(gst_base_rtp_audio_payload_handle_sample_based_buffer):
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_fixate):
Declare variables at the beginning of a block. Fixes #383195.
2006-12-09 15:12:38 +00:00
Tim-Philipp Müller
c90664d260 gst-libs/gst/interfaces/mixertrack.h: Add FIXME so we can add some padding here in 0.11
Original commit message from CVS:
* gst-libs/gst/interfaces/mixertrack.h:
Add FIXME so we can add some padding here in 0.11
2006-11-20 12:20:39 +00:00
Tim-Philipp Müller
23df03b763 gst-libs/gst/rtp/gstbasertpaudiopayload.h: Fix GstBaseRTPAudioPayload structure so the whole GObject inheritance busi...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
Fix GstBaseRTPAudioPayload structure so the whole GObject
inheritance business actually works (parent class instance structure
must always come first in the derived class instance structure).
2006-11-19 17:07:34 +00:00
David Schleef
12bfb95f3f configure.ac: Bump liboil requirement to 0.3.8.
Original commit message from CVS:
* configure.ac:
Bump liboil requirement to 0.3.8.
* gst-libs/gst/riff/riff-media.c:
Add Dirac fourcc.
* gst/videoscale/vs_image.h:
* gst/videoscale/vs_scanline.h:
Use liboil's stdint.h.
* gst/videotestsrc/videotestsrc.c:
Remove liboil related ifdef's, since they aren't needed now, and
won't work with future versions.
2006-11-14 23:34:19 +00:00
Jan David Mol
d24ad6ac0a Various gsize and gssize printf fixes. Fixes #372507.
Original commit message from CVS:
Patch by: Jan David Mol <j dot j dot d dot mol at tudelft dot nl>
* gst-libs/gst/riff/riff-read.c: (gst_riff_parse_strf_auds),
(gst_riff_parse_strf_iavs):
* gst/subparse/gstsubparse.c: (convert_encoding):
* gst/tcp/gstmultifdsink.c:
(gst_multi_fd_sink_handle_client_write):
* gst/tcp/gsttcp.c: (gst_tcp_socket_write), (gst_tcp_socket_read),
(gst_tcp_read_buffer), (gst_tcp_gdp_read_caps),
(gst_tcp_gdp_write_buffer), (gst_tcp_gdp_write_caps):
* gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_render):
* sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
(gst_ximagesink_ximage_new):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new):
Various gsize and gssize printf fixes. Fixes #372507.
2006-11-14 11:54:14 +00:00
Wim Taymans
0990cbf274 gst-libs/gst/audio/gstbaseaudiosink.*: Make the clock sync code more accurate wrt resampling and playback at differen...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_event), (gst_base_audio_sink_render):
* gst-libs/gst/audio/gstbaseaudiosink.h:
Make the clock sync code more accurate wrt resampling and playback
at different rates.
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_commit_full), (gst_ring_buffer_commit):
* gst-libs/gst/audio/gstringbuffer.h:
Use better algorithm to interpolate sample rates.
2006-11-13 17:30:17 +00:00
Stefan Kost
86de56e49d gst-libs/gst/riff/riff-media.c: add h263/h264 variants to the caps, Fixes #363118
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
(gst_riff_create_video_template_caps):
add h263/h264 variants to the caps, Fixes #363118
2006-11-07 07:22:32 +00:00
Tim-Philipp Müller
7298ebaa61 gst-libs/gst/audio/: Use g_strerror instead of strerror so we get UTF-8.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
* gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
Use g_strerror instead of strerror so we get UTF-8.
2006-11-06 18:24:59 +00:00
Wim Taymans
351622d028 gst-libs/gst/rtp/: Fix and activate base audio payloader.
Original commit message from CVS:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_init):
Fix and activate base audio payloader.
2006-10-31 10:49:19 +00:00
Tim-Philipp Müller
cc156bc97c gst-libs/gst/riff/riff-read.c: If strings in INFO chunk are not UTF-8, do something similar to what we do for ID3v1 t...
Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c: (freeform_string_to_utf8),
(gst_riff_parse_info):
If strings in INFO chunk are not UTF-8, do something similar to
what we do for ID3v1 tags: check a number of environment variables
(GST_AVI_TAG_ENCODING, GST_RIFF_TAG_ENCODING, GST_TAG_ENCODING) for
character sets to try, otherwise try the current locale and/or fall
back on ISO-8859-1. Fixes #360552.
2006-10-26 10:49:00 +00:00
Wim Taymans
1166abbc99 gst-libs/gst/audio/gstbaseaudiosink.*: Extract rate from the NEWSEGMENT event.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_event), (gst_base_audio_sink_render):
* gst-libs/gst/audio/gstbaseaudiosink.h:
Extract rate from the NEWSEGMENT event.
Use commit_full to also take rate adjustment into account when writing
samples to the ringbuffer.
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_commit_full), (gst_ring_buffer_commit),
(gst_ring_buffer_read):
* gst-libs/gst/audio/gstringbuffer.h:
Added _commit_full() to also take rate into account.
Use simple interpolation algorithm to resample audio.
API: gst_ring_buffer_commit_full()
* tests/examples/seek/scrubby.c: (speed_cb), (do_seek):
* tests/examples/seek/seek.c: (segment_done):
Don't try to seek with 0.0 rate, just pause instead.
Remove bogus debug line.
2006-10-18 13:42:49 +00:00
Tim-Philipp Müller
acba248480 gst-libs/gst/interfaces/tuner.c: Fix some function guards, add some more function guards.
Original commit message from CVS:
* gst-libs/gst/interfaces/tuner.c: (gst_tuner_list_channels),
(gst_tuner_set_channel), (gst_tuner_get_channel),
(gst_tuner_list_norms), (gst_tuner_set_norm), (gst_tuner_get_norm),
(gst_tuner_set_frequency), (gst_tuner_get_frequency),
(gst_tuner_signal_strength), (gst_tuner_find_norm_by_name),
(gst_tuner_find_channel_by_name):
Fix some function guards, add some more function guards.
2006-10-17 17:17:16 +00:00
Ville Syrjala
9b139e41fb gst-libs/gst/audio/gstbaseaudiosink.c: Don't crash when ringbuffer is not yet created.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_setcaps):
Don't crash when ringbuffer is not yet created.
Patch by: Ville Syrjala <ville dot syrjala at movial dot fi>
Fixes #361634.
* gst/playback/gstplaybasebin.c: (new_decoded_pad_full):
* gst/playback/gststreamselector.c:
(gst_stream_selector_request_new_pad):
Activate pads befre adding them to running elements.
2006-10-13 14:15:42 +00:00
Tim-Philipp Müller
e01b8a62ca gst-libs/gst/interfaces/mixer.c: Guard mixer interface functions against bogus arguments.
Original commit message from CVS:
* gst-libs/gst/interfaces/mixer.c: (gst_mixer_list_tracks),
(gst_mixer_set_volume), (gst_mixer_get_volume),
(gst_mixer_set_mute), (gst_mixer_set_option),
(gst_mixer_get_option), (gst_mixer_mute_toggled),
(gst_mixer_record_toggled), (gst_mixer_volume_changed),
(gst_mixer_option_changed):
Guard mixer interface functions against bogus arguments.
2006-10-13 08:57:52 +00:00
Sebastien Cote
014ce1511c gst-libs/gst/rtp/gstbasertppayload.c: Fix two small memory leaks (#361456).
Original commit message from CVS:
Patch by: Sebastien Cote  <sebas642 at yahoo.ca>
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
(gst_basertppayload_finalize):
Fix two small memory leaks (#361456).
2006-10-12 19:09:06 +00:00
Wim Taymans
98b1630fdc gst-libs/gst/riff/: Add WMS caps.
Original commit message from CVS:
* gst-libs/gst/riff/riff-ids.h:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
(gst_riff_create_audio_template_caps):
Add WMS caps.
2006-10-10 16:50:06 +00:00
Josep Torre Valles
4de10dacb6 ext/gnomevfs/: Fix URI interface implementation return type.
Original commit message from CVS:
2006-10-10  Zaheer Abbas Merali  <zaheerabbas at merali dot org>

Patch by: Josep Torre Valles <josep@fluendo.com>

* ext/gnomevfs/gstgnomevfssink.c:
* ext/gnomevfs/gstgnomevfssrc.c:
Fix URI interface implementation return type.
* ext/pango/gsttextoverlay.c: (gst_text_overlay_set_property):
Fix what looks like a copy/paste issue when assigning values.
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
(gst_audio_filter_template_get_type):
Cast to prevent Forte warnings.
* gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
Fix URI interface implementation return type.
gst_pad_query_position requires a signed integer pointer as
3rd parameter, GstClockTime is unsigned.
* gst/audioconvert/audioconvert.c:
Fix integer overflow when treated as signed.
* gst/audioresample/resample.c: (resample_add_input_data):
Cast to prevent warnings on Forte.
* gst/ffmpegcolorspace/imgconvert.c: (build_rgb_palette):
Fix integer overflow when treated as signed.
* gst/ffmpegcolorspace/imgconvert_template.h:
Fix integer overflow when treated as signed. RGBA_OUT shifts bits.
* gst/playback/gstdecodebin.c: (queue_filled_cb),
(cleanup_decodebin):
Who initialises a guint to -1!
Cast function pointers to prevent warnings on Forte.
* gst/playback/gstplaybasebin.c: (queue_deadlock_check),
(queue_threshold_reached):
Cast function pointers correctly to prevent warnings on Forte.
* gst/playback/gststreaminfo.c: (gst_stream_info_dispose):
Cast function pointers correctly to prevent warnings on Forte.
* gst/subparse/gstssaparse.c: (gst_ssa_parse_setcaps):
Obvious change to unsigned, 0xEF > max signed char.
* gst/tcp/gstmultifdsink.c: (get_buffers_max), (count_burst_unit):
GstClockTime is unsigned, initialise correctly.
* gst/tcp/gsttcp.c: (gst_tcp_socket_write):
Cast so pointer arithemetic doesn't cause warnings on Forte.
* gst/videorate/gstvideorate.c:
Use correct return value.
* tests/examples/seek/scrubby.c:
GstClockTime is unsigned, initialise correctly.
2006-10-10 12:49:03 +00:00
Tim-Philipp Müller
a336b4c10d gst-libs/gst/interfaces/xoverlay.c: Some more guards against invalid input.
Original commit message from CVS:
* gst-libs/gst/interfaces/xoverlay.c:
(gst_x_overlay_set_xwindow_id), (gst_x_overlay_expose):
Some more guards against invalid input.
2006-10-08 16:59:31 +00:00
Stefan Kost
ae6f56a20d gst-libs/gst/interfaces/xoverlay.c: Unref message in doc-example (spotted by Robert McQueen)
Original commit message from CVS:
* gst-libs/gst/interfaces/xoverlay.c:
Unref message in doc-example (spotted by Robert McQueen)
2006-10-06 19:20:53 +00:00
Michael Smith
0934b396d0 gst-libs/gst/floatcast/floatcast.h: Fix obviously-bogus macros; use the correct types.
Original commit message from CVS:
* gst-libs/gst/floatcast/floatcast.h:
Fix obviously-bogus macros; use the correct types.
2006-10-06 14:04:53 +00:00
Wim Taymans
07aaf7f948 gst-libs/gst/rtp/gstbasertpdepayload.c: Also call parent state change function to activate pads.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_change_state):
Also call parent state change function to activate pads.
* gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
(mpeg1_parse_header), (mpeg1_sys_type_find):
Add some more debug info in mpeg typefinding.
2006-10-06 13:34:46 +00:00
Tim-Philipp Müller
9e107d670a Printf format fixes.
Original commit message from CVS:
* ext/alsa/gstalsadeviceprobe.c:
(gst_alsa_device_property_probe_get_values):
* ext/alsa/gstalsasink.c: (set_hwparams):
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_elem_pad),
(gst_ogg_chain_new_stream), (gst_ogg_demux_read_chain):
* ext/ogg/gstoggmux.c: (gst_ogg_mux_send_headers),
(gst_ogg_mux_process_best_pad):
* ext/ogg/gstoggparse.c: (gst_ogg_parse_new_stream),
(gst_ogg_parse_chain):
* ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header):
* ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_setup),
(gst_vorbis_enc_buffer_check_discontinuous):
* ext/vorbis/vorbisparse.c: (vorbis_parse_src_query):
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_handle_track_seek):
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_push_full):
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
* gst/audioresample/resample.c: (resample_input_pushthrough):
* gst/playback/gstplaybasebin.c: (queue_out_of_data):
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_handle_clients):
* gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
(wavpack_type_find):
* gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_create):
* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
(gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new):
* tests/check/elements/volume.c: (GST_START_TEST):
Printf format fixes.
2006-10-05 15:55:21 +00:00
Philippe Kalaf
306ab03865 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Removed empty * between paragraphs
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
Removed empty * between paragraphs
2006-09-30 00:14:20 +00:00
Philippe Kalaf
5ba46c0866 gst-libs/gst/rtp/: Moved some documentation into .c file
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/rtp/README:
Moved some documentation into .c file
2006-09-29 23:50:53 +00:00
Wim Taymans
9945d7a468 gst-libs/gst/audio/gstbaseaudiosink.c: When we have a timestamp, we can still perform clipping.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
When we have a timestamp, we can still perform clipping.
When we have no clock, we must play the sample ASAP.
2006-09-28 15:08:15 +00:00
Wim Taymans
1980f16731 gst-libs/gst/audio/gstbaseaudiosink.c: Add some more info in a WARNING.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
Add some more info in a WARNING.
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
Handle PAUSE in create function, use new -core addition to
wait for playing. Fixes pausing and resuming capture from an
audiosrc.
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_commit),
(gst_ring_buffer_read):
Constify some more.
Caller supports interrupted reads now.
2006-09-27 13:52:14 +00:00
Wim Taymans
de735968c3 gst-libs/gst/rtp/gstbasertpdepayload.c: the source pad always uses fixed caps.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_init), (gst_base_rtp_depayload_push_full),
(gst_base_rtp_depayload_push_ts), (gst_base_rtp_depayload_process),
(gst_base_rtp_depayload_set_gst_timestamp):
the source pad always uses fixed caps.
2006-09-27 11:06:54 +00:00
Wim Taymans
7367722509 Added docs for the audio libs.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/gstaudioclock.c:
* gst-libs/gst/audio/gstaudioclock.h:
* gst-libs/gst/audio/gstaudiosink.c:
* gst-libs/gst/audio/gstaudiosink.h:
* gst-libs/gst/audio/gstaudiosrc.c:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
* gst-libs/gst/audio/gstbaseaudiosink.h:
* gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init):
* gst-libs/gst/audio/gstbaseaudiosrc.h:
* gst-libs/gst/audio/gstringbuffer.h:
Added docs for the audio libs.
2006-09-27 11:05:08 +00:00
Philippe Kalaf
214a128382 gst-libs/gst/rtp/gstbasertpaudiopayload.*: Moved AudioCodecType into priv
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
Moved AudioCodecType into priv
Renamed all gst_basertpaudiopayload to gst_base_rtp_audio_payload prefixes
2006-09-27 00:13:29 +00:00
Tim-Philipp Müller
787e58107c gst-libs/gst/tag/gstvorbistag.c: Parse dates that are followed by a time as well (#357532).
Original commit message from CVS:
* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
Parse dates that are followed by a time as well (#357532).
* tests/check/libs/tag.c: (test_vorbis_tags):
Add unit test for this.
2006-09-25 10:21:31 +00:00
Tim-Philipp Müller
77c1be94ad gst-libs/gst/Makefile.am: Need to build tag directory before cdda.
Original commit message from CVS:
* gst-libs/gst/Makefile.am:
Need to build tag directory before cdda.
2006-09-23 13:32:07 +00:00
Tim-Philipp Müller
ae45b7d755 Move GST_TAG_CDDA_* tags into libgsttag and make libgstcddabasesrc depend on libgsttag. This is required so we can ex...
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/cdda/Makefile.am:
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_base_init):
* gst-libs/gst/cdda/gstcddabasesrc.h:
* gst-libs/gst/tag/tag.h:
* gst-libs/gst/tag/tags.c: (gst_tag_register_tags_internal),
(gst_tag_register_musicbrainz_tags):
Move GST_TAG_CDDA_* tags into libgsttag and make libgstcddabasesrc
depend on libgsttag. This is required so we can extract/read tags like
DISCID without depending on libgstcddabasesrc (which used to register
them).
* gst-libs/gst/tag/gstvorbistag.c:
Add vorbiscomment mapping for CDDB_DISCID and MUSICBRAINZ_DISCID
tags (also see #347848).
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_metadata_set1):
Log vorbis comments we are actually writing. Const-ify array.
2006-09-23 13:21:07 +00:00
Young-Ho Cha
038488faa4 gst-libs/gst/riff/: Added MPEG-4 AAC and id and caps. Fixes #357289
Original commit message from CVS:
Patch by: Young-Ho Cha <ganadist at chollian dot net>
* gst-libs/gst/riff/riff-ids.h:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
(gst_riff_create_audio_template_caps):
Added MPEG-4 AAC and id and caps. Fixes #357289
Added WMA9 Lossless id.
2006-09-23 08:51:14 +00:00
Wim Taymans
7190c5f078 gst-libs/gst/rtp/gstbasertpdepayload.*: Small cleanups.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_finalize),
(gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
(gst_base_rtp_depayload_push_full),
(gst_base_rtp_depayload_push_ts), (gst_base_rtp_depayload_push),
(gst_base_rtp_depayload_process),
(gst_base_rtp_depayload_set_gst_timestamp),
(gst_base_rtp_depayload_queue_release):
* gst-libs/gst/rtp/gstbasertpdepayload.h:
Small cleanups.
Fix some leaks.
Refactored the process method and added methods to push from the process
vmethod.
Use _scale functions.
API: gst_base_rtp_depayload_push_ts
API: gst_base_rtp_depayload_push
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
timestamps are uint.
2006-09-22 14:13:34 +00:00
Stefan Kost
7b5c310767 gst-libs/gst/interfaces/xoverlay.c: Remove unused statement from doc example.
Original commit message from CVS:
* gst-libs/gst/interfaces/xoverlay.c:
Remove unused statement from doc example.
2006-09-22 11:59:00 +00:00
Stefan Kost
eba3881add gst-libs/gst/interfaces/videoorientation.c: Add since tags to new API docs, ChangeLog surgery (forgot API keyword in ...
Original commit message from CVS:
* gst-libs/gst/interfaces/videoorientation.c:
(gst_video_orientation_iface_init),
(gst_video_orientation_get_hflip),
(gst_video_orientation_get_vflip),
(gst_video_orientation_get_hcenter),
(gst_video_orientation_get_vcenter),
(gst_video_orientation_set_hflip),
(gst_video_orientation_set_vflip),
(gst_video_orientation_set_hcenter),
(gst_video_orientation_set_vcenter):
Add since tags to new API docs, ChangeLog surgery (forgot API keyword
in ChangeLog)
2006-09-21 13:49:47 +00:00
Wim Taymans
59b7c3104f gst-libs/gst/audio/gstringbuffer.c: Early morning compilation fix.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_samples_done), (gst_ring_buffer_commit),
(gst_ring_buffer_read):
Early morning compilation fix.
2006-09-21 05:12:18 +00:00
Wim Taymans
3edec5923c gst-libs/gst/riff/riff-media.c: Reorder the audio formats a bit for clarity.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
(gst_riff_create_audio_template_caps):
Reorder the audio formats a bit for clarity.
Detect and create caps for MSGSM and MSN (WAV49).
Fixes #356596.
* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
(gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new),
(gst_xvimagesink_get_xv_support), (gst_xvimagesink_show_frame):
Small cleanups, move error handling out of normal flow for clarity.
2006-09-19 10:23:03 +00:00
Stefan Kost
ce5339801d Add new interface to control video orientation (fixes #354908)
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs.types:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/interfaces/videoorientation.c:
(gst_video_orientation_get_type),
(gst_video_orientation_iface_init),
(gst_video_orientation_get_hflip),
(gst_video_orientation_get_vflip),
(gst_video_orientation_get_hcenter),
(gst_video_orientation_get_vcenter),
(gst_video_orientation_set_hflip),
(gst_video_orientation_set_vflip),
(gst_video_orientation_set_hcenter),
(gst_video_orientation_set_vcenter):
* gst-libs/gst/interfaces/videoorientation.h:
Add new interface to control video orientation (fixes #354908)
2006-09-18 15:59:39 +00:00
Stefan Kost
267a068e70 ext/gnomevfs/gstgnomevfssrc.c: Add docs about icydemux usage in connection with gnomevfssrc
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssrc.c:
Add docs about icydemux usage in connection with gnomevfssrc
* ext/libvisual/visual.c:
* ext/ogg/gstoggaviparse.c:
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggparse.c:
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
* gst-libs/gst/audio/gstaudiosink.c:
* gst-libs/gst/audio/gstaudiosrc.c:
* gst/audiorate/gstaudiorate.c:
More G_OBJECT macro fixing.
* gst/audiotestsrc/gstaudiotestsrc.h:
Fix wrong info in header due to copy & paste
2006-09-16 21:54:48 +00:00
Wim Taymans
65b1938b38 gst-libs/gst/audio/: Do the delay calculation in the source/sink base classes as this is specific for the capture/pla...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_get_time), (gst_base_audio_sink_callback):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_get_time), (gst_base_audio_src_fixate),
(gst_base_audio_src_get_times), (gst_base_audio_src_get_offset),
(gst_base_audio_src_create), (gst_base_audio_src_change_state):
Do the delay calculation in the source/sink base classes as this is
specific for the capture/playback mode.
Try to fixate a bit better, like round depth up to a multiple of 8
bigger than width.
Handle underruns correctly by marking DISCONT on buffers and adjusting
timestamps to handle the gap.
Set offset/offset_end correctly on buffers.
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_pause),
(gst_ring_buffer_samples_done), (gst_ring_buffer_commit),
(gst_ring_buffer_read):
Remove resync and underrun recovery from the ringbuffer.
Fix ringbuffer read code on under/overrun.
2006-09-15 14:53:44 +00:00
Wim Taymans
557b367295 configure.ac: We require 0.10.10.1 now because of _wait_preroll().
Original commit message from CVS:
* configure.ac:
We require 0.10.10.1 now because of _wait_preroll().
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
Use gst_base_sink_wait_preroll().
2006-09-15 09:13:50 +00:00
Thomas Vander Stichele
280e2ca67b releasing 0.10.10
Original commit message from CVS:
releasing 0.10.10
2006-09-14 20:09:19 +00:00
Wim Taymans
253831a549 gst-libs/gst/riff/riff-read.c: Protect public functions against bad input.
Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk),
(gst_riff_parse_chunk), (gst_riff_parse_file_header),
(gst_riff_parse_strh), (gst_riff_parse_strf_vids),
(gst_riff_parse_strf_auds), (gst_riff_parse_strf_iavs),
(gst_riff_parse_info):
Protect public functions against bad input.
Do some cleanups.
Fix documentation.
2006-08-22 16:31:47 +00:00
Tim-Philipp Müller
688cce0988 gst-libs/gst/riff/: Add voxware audio IDs (even if we can't play it) (#351795).
Original commit message from CVS:
* gst-libs/gst/riff/riff-ids.h:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Add voxware audio IDs (even if we can't play it) (#351795).
2006-08-22 15:50:36 +00:00
Tim-Philipp Müller
27c5a22fd7 gst-libs/gst/riff/riff-media.c: Const-ify some arrays and use G_N_ELEMENTS instead of wasting oodles of RAM on termin...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_template_caps),
(gst_riff_create_audio_template_caps),
(gst_riff_create_iavs_template_caps):
Const-ify some arrays and use G_N_ELEMENTS instead
of wasting oodles of RAM on terminator bits.
2006-08-22 15:11:52 +00:00
Tim-Philipp Müller
31d27aa08c And the same for _to_vorbiscomment_buffer(): allow id_data_len == 0 for speex.
Original commit message from CVS:
* gst-libs/gst/tag/gstvorbistag.c:
(gst_tag_list_to_vorbiscomment_buffer):
* tests/check/libs/tag.c: (GST_START_TEST):
And the same for _to_vorbiscomment_buffer(): allow
id_data_len == 0 for speex.
2006-08-22 08:27:07 +00:00
Tim-Philipp Müller
8c7c9ad6d4 gst-libs/gst/tag/gstvorbistag.c: Allow id_data_len == 0 (needed for vorbis comments in Speex files).
Original commit message from CVS:
* gst-libs/gst/tag/gstvorbistag.c:
(gst_tag_list_from_vorbiscomment_buffer):
Allow id_data_len == 0 (needed for vorbis comments in Speex files).
Also add some checks to make sure we don't memcmp() beyond the end of
vorbiscomment buffer if the ID to check for is larger than the buffer.
* tests/check/libs/tag.c: (GST_START_TEST):
Some more tests for gst_tag_list_from_vorbiscomment_buffer().
2006-08-21 18:34:46 +00:00
Wim Taymans
f212805058 docs/libs/: Don't try to GObject scan the netbuffer as it's not a GObject.
Original commit message from CVS:
* docs/libs/Makefile.am:
* docs/libs/gst-plugins-base-libs-sections.txt:
* docs/libs/gst-plugins-base-libs.types:
Don't try to GObject scan the netbuffer as it's not a GObject.
Fixes #351308.
* gst-libs/gst/netbuffer/gstnetbuffer.c:
* gst-libs/gst/netbuffer/gstnetbuffer.h:
Document GstNetBuffer.
2006-08-21 10:32:51 +00:00
Wim Taymans
cfb0252782 Document GstRTPBuffer.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_set_packet_len),
(gst_rtp_buffer_pad_to), (gst_rtp_buffer_get_payload_subbuffer),
(gst_rtp_buffer_get_payload_buffer):
* gst-libs/gst/rtp/gstrtpbuffer.h:
Document GstRTPBuffer.
Added function to efficiently strip payload headers.
API: gst_rtp_buffer_get_payload_subbuffer()
2006-08-18 16:43:26 +00:00
Tim-Philipp Müller
e7c6df131b gst-libs/gst/tag/gstvorbistag.c: Serialise unknown vorbis comments into GST_TAG_EXTENDED_COMMENT tags and deserialise...
Original commit message from CVS:
* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add),
(gst_tag_to_vorbis_comments):
Serialise unknown vorbis comments into GST_TAG_EXTENDED_COMMENT
tags and deserialise them properly as well (#351768).
Add some more gtk-doc blurbs and also some g_return_if_fail().
* tests/check/libs/tag.c: (GST_START_TEST),
(back_to_vorbis_comments), (taglists_are_equal), (tag_suite):
More tests.
2006-08-17 16:52:06 +00:00
Wim Taymans
c973b2bf3e ext/ogg/: Added ogg-in-avi parser element. Fixes #140139.
Original commit message from CVS:
* ext/ogg/Makefile.am:
* ext/ogg/gstogg.c: (plugin_init):
* ext/ogg/gstoggaviparse.c: (gst_ogg_avi_parse_get_type),
(gst_ogg_avi_parse_base_init), (gst_ogg_avi_parse_class_init),
(gst_ogg_avi_parse_init), (gst_ogg_avi_parse_finalize),
(gst_ogg_avi_parse_setcaps), (gst_ogg_avi_parse_event),
(gst_ogg_avi_parse_push_packet), (gst_ogg_avi_parse_chain),
(gst_ogg_avi_parse_change_state), (gst_ogg_avi_parse_plugin_init):
Added ogg-in-avi parser element. Fixes #140139.
* ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page):
Fixed a bug in oggdemux debug code.
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
(gst_riff_create_audio_template_caps):
Recognise Ogg in the AVI extensible wave format.
2006-08-17 15:43:40 +00:00
Tim-Philipp Müller
75ecc060b1 gst-libs/gst/cdda/gstcddabasesrc.c: Make buffer durations add up (duration should be next_ts-ts for perfect streams)....
Original commit message from CVS:
* gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
Make buffer durations add up (duration should be next_ts-ts for
perfect streams). Fixes CD ripping to Ogg/Vorbis with vorbisenc
from CVS.
* tests/check/libs/cddabasesrc.c: (gst_cd_foo_src_close),
(test_buffer_timestamps), (cddabasesrc_suite):
Add unit test for the above.
* tests/check/Makefile.am:
Don't know why cddabasesrc test was in VALGRIND_TO_FIX, remove
to see what happens.
2006-08-17 10:00:00 +00:00
Tim-Philipp Müller
6aeb8149dd API: add gst_tag_parse_extended_comment() (#351426).
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/tag/tag.h:
* gst-libs/gst/tag/tags.c: (gst_tag_parse_extended_comment):
API: add gst_tag_parse_extended_comment() (#351426).
* tests/check/Makefile.am:
* tests/check/libs/.cvsignore:
* tests/check/libs/tag.c: (GST_START_TEST), (tag_suite), (main):
Add unit test for gst_tag_parse_extended_comment().
2006-08-16 11:28:57 +00:00
Young-Ho Cha
33e0a62c5c gst-libs/gst/riff/: Detect DTS audio streams (#350157).
Original commit message from CVS:
Patch by: Young-Ho Cha  <ganadist at chollian net>
* gst-libs/gst/riff/riff-ids.h:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
(gst_riff_create_audio_template_caps):
Detect DTS audio streams (#350157).
2006-08-07 08:26:03 +00:00
Tim-Philipp Müller
ea41bfefd7 ext/alsa/: Add support for cards that (only) do more than 8 channels, like the Delta 44 (#345188).
Original commit message from CVS:
* ext/alsa/gstalsa.c: (caps_add_channel_configuration),
(gst_alsa_detect_channels):
* ext/alsa/gstalsasink.c:
Add support for cards that (only) do more than 8 channels,
like the Delta 44 (#345188).
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions):
* gst-libs/gst/audio/multichannel.h:
API: add GST_AUDIO_CHANNEL_POSITION_NONE, which stands for an
unspecified channel position and cannot be combined with any
of the other audio channel positions; adjust position layout
checks accordingly (#345188).
2006-08-03 14:16:06 +00:00
Marcel Moreaux
86d007a553 gst-libs/gst/rtp/gstbasertpdepayload.*: Handle RTP sequence number rollover.
Original commit message from CVS:
Patch by: Marcel Moreaux <marcelm at luon dot net>
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_add_to_queue):
* gst-libs/gst/rtp/gstbasertpdepayload.h:
Handle RTP sequence number rollover.
Disable jitterbuffer by default.
2006-08-02 17:03:29 +00:00
Kai Vehmanen
2a872ad81d gst-libs/gst/rtp/gstbasertpdepayload.c: Don't send multiple newsegments with different formats.
Original commit message from CVS:
patch by: Kai Vehmanen <kv2004 eca cx>
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_chain),
(gst_base_rtp_depayload_handle_sink_event),
(gst_base_rtp_depayload_change_state):
Don't send multiple newsegments with different formats.
Fixes #348677.
2006-07-27 10:52:52 +00:00
Wim Taymans
d5a10b05c2 gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to align a sample to an unknown value.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
Don't try to align a sample to an unknown value.
2006-07-24 16:47:10 +00:00
Wim Taymans
f3ae89426a gst-libs/gst/audio/gstbaseaudiosink.c: When the audio clock is slaved to another clock, never try to align samples bu...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_provide_clock), (gst_base_audio_sink_render):
When the audio clock is slaved to another clock, never try to align
samples but trust the rate interpolation algorithm.
2006-07-24 15:14:17 +00:00
Wim Taymans
19cd03c607 ext/alsa/gstalsasink.c: Don't try to calculate silence samples, base class does this much better now.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
Don't try to calculate silence samples, base class does this much
better now.
* gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
(gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps),
(gst_ring_buffer_acquire):
Calculate silence samples correctly.
* gst-libs/gst/audio/gstringbuffer.h:
Add _CAST macro.
2006-07-24 14:34:42 +00:00
Wim Taymans
843202b51c gst-libs/gst/audio/gstaudiosink.c: Fix leak.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
(gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire),
(gst_audioringbuffer_release), (gst_audioringbuffer_stop):
Fix leak.
Avoid type casting when we can.
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_dispose):
Fix mem leak.
2006-07-21 10:43:54 +00:00
Stefan Kost
d63372f3bb gst-libs/gst/riff/riff-ids.h: Add 'fact' chunk id
Original commit message from CVS:
* gst-libs/gst/riff/riff-ids.h:
Add 'fact' chunk id
2006-07-20 05:56:48 +00:00
Wim Taymans
01402bc9e3 gst-libs/gst/rtp/gstbasertpdepayload.c: Don't assert when not negotiated but post a meaningfull error message. Fixes ...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_chain),
(gst_base_rtp_depayload_change_state):
Don't assert when not negotiated but post a meaningfull
error message. Fixes #347918.
* gst-libs/gst/rtp/gstbasertppayload.c:
Add comment about better default MTU size.
* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data):
Small cleanups, start docs.
2006-07-19 18:20:43 +00:00
Tim-Philipp Müller
f8e63966ad gst-libs/gst/tag/gstid3tag.c: Add TSSE <=> GST_TAG_ENCODER mapping (see #347898).
Original commit message from CVS:
* gst-libs/gst/tag/gstid3tag.c:
Add TSSE <=> GST_TAG_ENCODER mapping (see #347898).
* gst-libs/gst/tag/gsttageditingprivate.h:
* gst-libs/gst/tag/gstvorbistag.c:
Some more random const-ifications.
2006-07-19 12:25:00 +00:00
Stefan Kost
794586031e gst-libs/gst/riff/: add comment what those 16 bytes in struct _gst_riff_strh according to one avi-dumper are
Original commit message from CVS:
* gst-libs/gst/riff/riff-ids.h:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_template_caps):
Add more FOURCCs (sort list to make stuff easier to find),
add comment what those 16 bytes in struct _gst_riff_strh according to
one avi-dumper are
2006-07-18 19:48:48 +00:00
Tim-Philipp Müller
a56652b204 gst-libs/gst/audio/multichannel.c: Const-ify two arrays.
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions),
(gst_audio_fixate_channel_positions):
Const-ify two arrays.
2006-07-17 13:48:10 +00:00
Wim Taymans
bbe88d8dab gst-libs/gst/rtp/gstbasertpdepayload.c: Fix 99% cpu load by waiting for absolute times on the clock. Fixes #347300.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_wait),
(gst_base_rtp_depayload_change_state),
(gst_base_rtp_depayload_set_property),
(gst_base_rtp_depayload_get_property):
Fix 99% cpu load by waiting for absolute times on the
clock. Fixes #347300.
2006-07-14 17:56:59 +00:00
Wim Taymans
a0354a5b96 gst-libs/gst/audio/gstbaseaudiosrc.c: Don't try to post an error message when setting the clock fails as this can hap...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_set_clock),
(gst_base_audio_src_check_get_range), (gst_base_audio_src_create):
Don't try to post an error message when setting the clock fails
as this can happen when adding an element to a bin which will then
deadlock. Fixes #347296.
2006-07-12 13:24:19 +00:00
Wim Taymans
ccee48bb85 Revert last two changes that broke the freeze.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
* gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
(gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps):
Revert last two changes that broke the freeze.
2006-07-12 11:28:37 +00:00
Wim Taymans
46d86d8005 gst-libs/gst/audio/gstringbuffer.c: Calculate correct silence samples so we don't fill our ringbuffer with noise.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
(gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps):
Calculate correct silence samples so we don't fill our ringbuffer
with noise.
2006-07-12 10:58:42 +00:00
Wim Taymans
fa5dacc998 gst-libs/gst/audio/gstbaseaudiosink.c: Use gobject_class instead of G_OBJECT_CLASS (klass)
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init),
(gst_base_audio_sink_provide_clock):
Use gobject_class instead of G_OBJECT_CLASS (klass)
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init), (gst_base_audio_src_init),
(gst_base_audio_src_set_clock), (gst_base_audio_src_provide_clock),
(gst_base_audio_src_get_time),
(gst_base_audio_src_check_get_range), (gst_base_audio_src_create),
(gst_base_audio_src_create_ringbuffer):
Fix latency and buffer-time constants and properties ala basesink.
Implement pull based scheduling. Fixes #346527.
Set default blocksize in GstBaseSrc to 0, we default to pushing out
one segment.
Refuse slaving to another clock instead of silently not working.
Only provide a clock when we are actually able to do so.
Various small cleanups and compiler hints.
2006-07-06 15:54:50 +00:00
Tim-Philipp Müller
31667522ec gst-libs/gst/riff/riff-media.c: Recognise 'WMVA' video codec fourcc (#345879).
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
Recognise 'WMVA' video codec fourcc (#345879).
2006-06-30 12:04:51 +00:00
Philippe Kalaf
7e52276a83 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Fixed nasty memory leak
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
Fixed nasty memory leak
2006-06-29 12:21:06 +00:00
Tim-Philipp Müller
114a273f27 Use GST_DEBUG_CATEGORY_STATIC where possible (#342503).
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
* gst-libs/gst/rtp/gstbasertppayload.c:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/tcp/gstmultifdsink.c:
* gst/tcp/gsttcpclientsink.c:
* gst/tcp/gsttcpclientsrc.c:
* gst/tcp/gsttcpserversink.c:
* gst/tcp/gsttcpserversrc.c:
* gst/videorate/gstvideorate.c:
* gst/videotestsrc/gstvideotestsrc.c:
* sys/v4l/gstv4ljpegsrc.c:
* sys/v4l/gstv4lmjpegsink.c:
* sys/v4l/gstv4lsrc.c:
* tests/examples/seek/scrubby.c:
* tests/examples/seek/seek.c:
Use GST_DEBUG_CATEGORY_STATIC where possible (#342503).
2006-06-23 09:53:09 +00:00
Tim-Philipp Müller
96331d17e5 gst-libs/gst/tag/tags.c: Second field in GEnumValue shouldn't be a description, but a stringified version of the enum...
Original commit message from CVS:
* gst-libs/gst/tag/tags.c: (register_tag_image_type_enum):
Second field in GEnumValue shouldn't be a description,
but a stringified version of the enum value.
2006-06-22 12:13:31 +00:00
Tim-Philipp Müller
96326bff9d gst-libs/gst/tag/tag.h: Oops, make that 'Since: 0.10.9'.
Original commit message from CVS:
* gst-libs/gst/tag/tag.h:
Oops, make that 'Since: 0.10.9'.
2006-06-22 11:01:41 +00:00
Tim-Philipp Müller
f8f78cb023 API: add GstTagImageType enum to describe images contained in image tags (#345641).
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/tag/tag.h:
* gst-libs/gst/tag/tags.c: (register_tag_image_type_enum),
(gst_tag_image_type_get_type):
API: add GstTagImageType enum to describe images contained
in image tags (#345641).
2006-06-22 10:55:05 +00:00
Young-Ho Cha
f1392c148e gst-libs/gst/riff/riff-read.c: Parse extra data better, apparently it's right behind the normal strf header size. Fix...
Original commit message from CVS:
Patch by: Young-Ho Cha <ganadist at chollian dot net>
* gst-libs/gst/riff/riff-read.c: (gst_riff_parse_strf_vids):
Parse extra data better, apparently it's right behind
the normal strf header size. Fixes #343500.
2006-06-16 13:59:29 +00:00
Tim-Philipp Müller
5288476e31 Remove GST_CDDA_TAG_TRACK_TAGS again, it is #ifdef 0'ed out in the header file and shouldn't be listed in the docs.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/cdda/gstcddabasesrc.h:
Remove GST_CDDA_TAG_TRACK_TAGS again, it is #ifdef 0'ed
out in the header file and shouldn't be listed in the docs.
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
Fix it so that it doesn't crash in the debug statement.
2006-06-16 10:20:10 +00:00
Stefan Kost
cade791150 docs/libs/: add remaining symbols into correct setions
Original commit message from CVS:
* docs/libs/Makefile.am:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* docs/libs/gst-plugins-base-libs.types:
add remaining symbols into correct setions
* gst-libs/gst/audio/gstringbuffer.c:
fix incomplete docs
* gst-libs/gst/audio/gstringbuffer.h:
comment out not yet implemented function
* gst-libs/gst/floatcast/floatcast.h:
* gst-libs/gst/netbuffer/gstnetbuffer.c:
add short descriptions
* gst-libs/gst/interfaces/propertyprobe.c:
fix return value docs
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
simplify debug logging
* gst-libs/gst/riff/riff-read.h:
sync function prototype and docs
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
remove left over symbol
2006-06-16 10:02:25 +00:00
Tim-Philipp Müller
7c145357ea gst-libs/gst/riff/riff-ids.h: Add IDs for 'bext' chunks (see #343837).
Original commit message from CVS:
* gst-libs/gst/riff/riff-ids.h:
Add IDs for 'bext' chunks (see #343837).
2006-06-14 08:17:45 +00:00
Thomas Vander Stichele
51ca8fe3e1 move last template doc snippets to source code and delete them
Original commit message from CVS:
* docs/libs/tmpl/gstaudio.sgml:
* docs/libs/tmpl/gstcolorbalance.sgml:
* docs/libs/tmpl/gstmixer.sgml:
* docs/libs/tmpl/gstringbuffer.sgml:
* docs/libs/tmpl/gsttuner.sgml:
* docs/libs/tmpl/gstxoverlay.sgml:
* gst-libs/gst/audio/audio.c:
* gst-libs/gst/audio/gstringbuffer.c:
* gst-libs/gst/interfaces/colorbalance.c:
* gst-libs/gst/interfaces/mixer.c:
* gst-libs/gst/interfaces/tuner.c:
* gst-libs/gst/interfaces/xoverlay.c:
move last template doc snippets to source code and delete them
2006-06-07 11:03:03 +00:00
Jan Schmidt
45e06fe704 gst-libs/gst/audio/: Document better the fact that latency_time and buffer_time are values stored in microseconds, an...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init), (gst_base_audio_sink_setcaps):
* gst-libs/gst/audio/gstbaseaudiosink.h:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps),
(gst_ring_buffer_samples_done):
* gst-libs/gst/audio/gstringbuffer.h:
Document better the fact that latency_time and buffer_time are values
stored in microseconds, and not the usual GStreamer nanoseconds.
Change the variables (compatibly) that store them from GstClockTime
to guint64 to make it more clear that they're not storing clock times.
Also, remove the bogus property description that says the user can
specify -1 to get the default value, since that's never been the case.
When computing the default segment size for the ring buffer, make it
an integer number of samples.
When the sub-class indicates a delay greater than the number of
samples we've written return 0 from the audio sink get_time method.
2006-06-03 21:06:49 +00:00
Stefan Kost
131fb86b4b Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
Original commit message from CVS:
* ext/alsa/gstalsamixerelement.h:
* ext/alsa/gstalsamixeroptions.h:
* ext/alsa/gstalsamixertrack.h:
* ext/gnomevfs/gstgnomevfssink.h:
* ext/gnomevfs/gstgnomevfssrc.h:
* ext/theora/gsttheoradec.h:
* ext/theora/gsttheoraenc.h:
* ext/theora/gsttheoraparse.h:
* ext/vorbis/vorbisparse.h:
* gst-libs/gst/audio/gstaudioclock.h:
* gst-libs/gst/audio/gstaudiofilter.h:
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
* gst/audioconvert/gstaudioconvert.h:
* gst/audioresample/gstaudioresample.h:
* gst/audiotestsrc/gstaudiotestsrc.h:
* gst/ffmpegcolorspace/gstffmpegcolorspace.h:
* gst/playback/gststreamselector.h:
* gst/tcp/gstmultifdsink.h:
* gst/tcp/gsttcpclientsink.h:
* gst/tcp/gsttcpclientsrc.h:
* gst/tcp/gsttcpserversink.h:
* gst/tcp/gsttcpserversrc.h:
* gst/videorate/gstvideorate.h:
* gst/videoscale/gstvideoscale.h:
* gst/videotestsrc/gstvideotestsrc.h:
* gst/volume/gstvolume.h:
* sys/v4l/gstv4ljpegsrc.h:
* sys/v4l/gstv4lmjpegsink.h:
* sys/v4l/gstv4lmjpegsrc.h:
* sys/v4l/gstv4lsrc.h:
* sys/ximage/ximagesink.h:
* sys/xvimage/xvimagesink.h:
* tests/old/testsuite/alsa/sinesrc.h:
Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
2006-06-01 19:19:51 +00:00
Philippe Kalaf
0e710f94cc gst-libs/gst/rtp/gstbasertpaudiopayload.c: 80 line columns
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
80 line columns
Removed redundant floor()
2006-05-19 17:57:56 +00:00
Tim-Philipp Müller
25d29c4f09 gst-libs/gst/riff/riff-read.c: On second thought, just skip JUNK chunks automatically, so the caller doesn't have to ...
Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
On second thought, just skip JUNK chunks automatically, so
the caller doesn't have to handle this. Fixes #342345.
Also, return GST_FLOW_UNEXPECTED if we get a short read,
not GST_FLOW_ERROR.
2006-05-19 15:00:43 +00:00
Tim-Philipp Müller
3ef484fe2b gst-libs/gst/riff/riff-read.c: Don't bail out on JUNK chunks with a size of 0 (would try to pull_range 0 bytes before...
Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
Don't bail out on JUNK chunks with a size of 0 (would try to
pull_range 0 bytes before, which sources don't like too much).
See #342345.
2006-05-19 13:37:55 +00:00
Philippe Kalaf
8675bc89e4 gst-libs/gst/rtp/README: Some new documentation
Original commit message from CVS:
2006-05-18 Philippe Kalaf <philippe.kalaf@collabora.co.uk>

* gst-libs/gst/rtp/README:
Some new documentation
* gst-libs/gst/rtp/gstrtpbuffer.h:
Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
New RTP audio base payloader class. Supports frame or sample based codecs.
Not enabled in Makefile.am until approved.
2006-05-18 23:00:02 +00:00
Tim-Philipp Müller
10d35563dd gst-libs/gst/audio/multichannel.c: It's okay to have caps with channels=1 and a channel position different from GST_A...
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions):
It's okay to have caps with channels=1 and a channel position
different from GST_AUDIO_CHANNEL_POSITION_FRONT_MONO
(deinterleavers might want to keep the position in the caps,
so that they can be re-interleaved again properly later).
Leave check for unexpected 2-channel layouts intact for now.
2006-05-16 17:34:14 +00:00
Jan Schmidt
34db0838be Fix integer overflow problem with pixel-aspect-ratio calculations in videoscale and xvimagesink (#341542)
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/video/video.c: (gst_video_calculate_display_ratio):
* gst-libs/gst/video/video.h:
* gst/videoscale/Makefile.am:
* gst/videoscale/gstvideoscale.c: (gst_video_scale_fixate_caps):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps):
* tests/check/Makefile.am:
* tests/check/libs/video.c: (GST_START_TEST), (video_suite),
(main):
Fix integer overflow problem with pixel-aspect-ratio calculations
in videoscale and xvimagesink (#341542)
2006-05-12 21:30:00 +00:00
Tim-Philipp Müller
2f9b081b9f gst-libs/gst/tag/gstid3tag.c: Map GST_IMAGE_TAG to and from ID2v2 APIC frames (#341557).
Original commit message from CVS:
* gst-libs/gst/tag/gstid3tag.c:
Map GST_IMAGE_TAG to and from ID2v2 APIC frames (#341557).
2006-05-12 16:56:52 +00:00
Tim-Philipp Müller
d8965c30fb Const-ify GEnumValue and GFlagsValue arrays. Use
Original commit message from CVS:
* ext/pango/gsttextoverlay.c: (gst_text_overlay_valign_get_type),
(gst_text_overlay_halign_get_type),
(gst_text_overlay_wrap_mode_get_type):
* ext/theora/theoradec.c: (theora_handle_type_packet),
(theora_handle_data_packet):
* ext/theora/theoraenc.c: (gst_border_mode_get_type),
(theora_enc_sink_setcaps), (theora_enc_chain):
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_mode_get_type):
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audiostestsrc_wave_get_type):
* gst/playback/gststreaminfo.c: (gst_stream_type_get_type):
* gst/tcp/gstfdset.c: (gst_fdset_mode_get_type):
* gst/tcp/gstmultifdsink.c: (gst_recover_policy_get_type),
(gst_sync_method_get_type), (gst_unit_type_get_type),
(gst_client_status_get_type):
* gst/videoscale/gstvideoscale.c:
(gst_video_scale_method_get_type):
* gst/videotestsrc/gstvideotestsrc.c:
(gst_video_test_src_pattern_get_type):
* gst/videotestsrc/videotestsrc.c: (paint_setup_I420),
(paint_setup_YV12), (paint_setup_YUY2), (paint_setup_UYVY),
(paint_setup_YVYU), (paint_setup_IYU2), (paint_setup_Y41B),
(paint_setup_Y42B), (paint_setup_Y800), (paint_setup_YVU9),
(paint_setup_YUV9), (paint_setup_RGB888), (paint_setup_BGR888),
(paint_setup_RGB565), (paint_setup_xRGB1555):
Const-ify GEnumValue and GFlagsValue arrays. Use
GST_ROUND_UP_* macros instead of home-made ones.
2006-05-09 19:24:46 +00:00
Tim-Philipp Müller
308f541bf4 gst-libs/gst/riff/riff-media.c: Add SEDG (Samsung MPEG-4) fourcc.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
Add SEDG (Samsung MPEG-4) fourcc.
2006-05-09 11:37:22 +00:00
Thomas Vander Stichele
cccfb086bc doc reparagraphing and DEBUG_FUNCPTRing
Original commit message from CVS:
doc reparagraphing and DEBUG_FUNCPTRing
2006-05-08 15:51:15 +00:00
Thomas Vander Stichele
bc510bf9b8 some RTP debug
Original commit message from CVS:
some RTP debug
2006-05-02 06:33:54 +00:00
Tim-Philipp Müller
b981f0999b gst-libs/gst/riff/riff-media.c: Protect against unexpected NULL strf_data buffer.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Protect against unexpected NULL strf_data buffer.
2006-04-29 16:25:58 +00:00
Thomas Vander Stichele
6e3506b116 gst-libs/gst/cdda/gstcddabasesrc.c: compile fix; strtol() needs <stdlib.h>
Original commit message from CVS:

* gst-libs/gst/cdda/gstcddabasesrc.c:
compile fix; strtol() needs <stdlib.h>
2006-04-29 01:10:34 +00:00
Stefan Kost
e972defd3e make GstElementDetails const
Original commit message from CVS:
* ext/alsa/gstalsamixerelement.c:
* ext/alsa/gstalsasrc.c:
* ext/cdparanoia/gstcdparanoiasrc.c:
* ext/gnomevfs/gstgnomevfssink.c:
* ext/gnomevfs/gstgnomevfssrc.c:
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggparse.c:
* ext/ogg/gstogmparse.c:
* ext/pango/gstclockoverlay.c:
* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextrender.c:
* ext/pango/gsttimeoverlay.c:
* ext/theora/theoradec.c:
* ext/theora/theoraenc.c:
* ext/vorbis/vorbisdec.c:
* ext/vorbis/vorbisenc.c:
* gst-libs/gst/audio/gstaudiofilter.c:
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
* gst/audioconvert/gstaudioconvert.c:
* gst/audiorate/gstaudiorate.c:
* gst/audioresample/gstaudioresample.c:
* gst/audiotestsrc/gstaudiotestsrc.c:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/playback/gstdecodebin.c:
* gst/playback/gstplaybin.c:
* gst/playback/gststreamselector.c:
* gst/subparse/gstsubparse.c:
* gst/tcp/gstmultifdsink.c:
* gst/tcp/gsttcpclientsink.c:
* gst/tcp/gsttcpclientsrc.c:
* gst/tcp/gsttcpserversink.c:
* gst/tcp/gsttcpserversrc.c:
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
* gst/videorate/gstvideorate.c:
* gst/videoscale/gstvideoscale.c:
* gst/videotestsrc/gstvideotestsrc.c:
* gst/volume/gstvolume.c:
* sys/v4l/gstv4ljpegsrc.c:
* sys/v4l/gstv4lmjpegsink.c:
* sys/v4l/gstv4lmjpegsrc.c:
* sys/v4l/gstv4lsrc.c:
* sys/ximage/ximagesink.c:
* sys/xvimage/xvimagesink.c:
* tests/check/libs/cddabasesrc.c:
make GstElementDetails const
2006-04-28 19:46:37 +00:00
Wim Taymans
102b79e46e gst-libs/gst/audio/gstbaseaudiosink.c: patch to make timestamp checking more tollerant to rounding errors given that ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain), (gst_base_audio_sink_event),
(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
patch to make timestamp checking more tollerant to rounding
errors given that real discontinuities are to be marked on
buffers. Fixes some asf files and #338778.
Also avoid some crashers when we receive an event in the
NULL state.
2006-04-28 15:08:09 +00:00
Wim Taymans
04754176a6 gst-libs/gst/audio/gstringbuffer.c: Check arguments passed to public functions instead of crashing.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_set_callback), (gst_ring_buffer_acquire),
(gst_ring_buffer_release), (gst_ring_buffer_is_acquired),
(gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
(gst_ring_buffer_pause), (gst_ring_buffer_stop),
(gst_ring_buffer_delay), (gst_ring_buffer_samples_done),
(gst_ring_buffer_set_sample), (gst_ring_buffer_clear_all),
(gst_ring_buffer_commit), (gst_ring_buffer_read),
(gst_ring_buffer_prepare_read), (gst_ring_buffer_advance),
(gst_ring_buffer_clear), (gst_ring_buffer_may_start):
Check arguments passed to public functions instead of
crashing.
2006-04-28 14:48:11 +00:00
Wim Taymans
c068425b38 gst-libs/gst/audio/gstbaseaudiosrc.c: GstBaseAudioSrc must be live or it does not work.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
(gst_base_audio_src_get_time), (gst_base_audio_src_create):
GstBaseAudioSrc must be live or it does not work.
* gst-libs/gst/audio/gstaudiosrc.c: (gst_audio_src_init):
Don't set live to TRUE as this is the default in the parentclass.
2006-04-28 14:37:46 +00:00
Tim-Philipp Müller
d414feba1d gst-libs/gst/riff/riff-media.c: add another H264 fourcc. Fixes #339047.
Original commit message from CVS:

Patch by: Tim-Philipp Müller  <tim at centricular dot net>

* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
add another H264 fourcc.  Fixes #339047.
2006-04-22 21:19:06 +00:00
Thomas Vander Stichele
6f18aadd6b reverting rtp patches to fix freeze break on -base as explained on the list
Original commit message from CVS:
reverting rtp patches to fix freeze break on -base as explained on the list
2006-04-13 09:26:27 +00:00
Philippe Kalaf
a916af7c48 gst-libs/gst/rtp/gstrtpbuffer.h: Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
Original commit message from CVS:
2006-04-12 Philippe Kalaf <philippe.kalaf@collabora.co.uk>

* gst-libs/gst/rtp/gstrtpbuffer.h:
Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
New RTP audio base payloader class. Supports frame or sample based codecs
2006-04-13 03:55:12 +00:00
Antoine Tremblay
5c7a047016 gst-libs/gst/rtp/gstbasertpdepayload.c: Fix some memory leaks: on finalize, free buffers left in the queue before des...
Original commit message from CVS:
Patch by: Antoine Tremblay  <hexa00 at gmail dot com>
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_finalize), (gst_base_rtp_depayload_push):
Fix some memory leaks: on finalize, free buffers left in the queue
before destroying the queue; in _push(), unref rtp_buf even if
the process vfunc returned a NULL buffer as output buffer (#337548);
demote some recuring debug messages to LOG level.
2006-04-11 17:31:29 +00:00
Wim Taymans
35058f78c1 gst-libs/gst/audio/gstbaseaudiosink.c: Starting the ringbuffer when we did not acquire it can cause a deadlock, is po...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_event):
Starting the ringbuffer when we did not acquire it can cause
a deadlock, is pointless and causes nasty things for
subclasses.
Fixes gst-launch audiotestsrc num-buffers=0 ! alsasink.
2006-04-10 17:05:46 +00:00
Stefan Kost
0afac375b4 Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
Original commit message from CVS:
* ext/alsa/gstalsamixeroptions.c:
(gst_alsa_mixer_options_class_init):
* ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_class_init):
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_class_init):
* ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init):
* ext/ogg/gstoggparse.c: (gst_ogg_parse_class_init):
* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_class_init):
* gst-libs/gst/audio/gstaudiofilter.c:
(gst_audio_filter_class_init):
* gst-libs/gst/audio/gstaudiosink.c:
(gst_audioringbuffer_class_init):
* gst-libs/gst/audio/gstaudiosrc.c:
(gst_audioringbuffer_class_init):
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_class_init):
* gst-libs/gst/interfaces/colorbalancechannel.c:
(gst_color_balance_channel_class_init):
* gst-libs/gst/interfaces/mixeroptions.c:
(gst_mixer_options_class_init):
* gst-libs/gst/interfaces/mixertrack.c:
(gst_mixer_track_class_init):
* gst-libs/gst/interfaces/tunerchannel.c:
(gst_tuner_channel_class_init):
* gst-libs/gst/interfaces/tunernorm.c: (gst_tuner_norm_class_init):
* gst-libs/gst/netbuffer/gstnetbuffer.c:
(gst_netbuffer_class_init):
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_class_init):
* gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init):
* gst/playback/gstplaybin.c: (gst_play_bin_class_init):
* gst/playback/gststreaminfo.c: (gst_stream_info_class_init):
* gst/playback/gststreamselector.c:
(gst_stream_selector_class_init):
* gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
* gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
* sys/v4l/gstv4lcolorbalance.c:
(gst_v4l_color_balance_channel_class_init):
* sys/v4l/gstv4ljpegsrc.c: (gst_v4ljpegsrc_class_init):
* sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init):
* sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init):
* sys/v4l/gstv4ltuner.c: (gst_v4l_tuner_channel_class_init),
(gst_v4l_tuner_norm_class_init):
* sys/ximage/ximagesink.c: (gst_ximagesink_class_init):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
* tests/old/testsuite/alsa/sinesrc.c: (sinesrc_class_init):
Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
2006-04-08 21:02:53 +00:00
Stefan Kost
1a2642a1d2 Fix broken GObject macros
Original commit message from CVS:
* ext/pango/gsttextrender.h:
* gst-libs/gst/audio/gstaudiosink.h:
* gst-libs/gst/audio/gstaudiosrc.h:
* gst-libs/gst/audio/gstbaseaudiosink.h:
* gst-libs/gst/audio/gstbaseaudiosrc.h:
* gst-libs/gst/audio/gstringbuffer.h:
* gst-libs/gst/rtp/gstbasertpdepayload.h:
* gst-libs/gst/rtp/gstbasertppayload.h:
* gst-libs/gst/video/gstvideofilter.h:
* gst-libs/gst/video/gstvideosink.h:
* gst/playback/gstplaybasebin.h:
* gst/tcp/gstmultifdsink.h:
* sys/v4l/gstv4lelement.h:
Fix broken GObject macros
2006-04-08 18:09:17 +00:00
Stefan Kost
2d826700fa Add docs for adder, use GST_ELEMENT_DETAILS macro, define GstElementDetails at the top
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* ext/cdparanoia/gstcdparanoiasrc.c:
* ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_base_init),
(gst_gnome_vfs_sink_class_init):
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_base_init):
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_base_init):
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggparse.c: (gst_ogg_parse_base_init):
* ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init),
(gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init):
* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextrender.c:
* ext/theora/theoradec.c:
* ext/theora/theoraenc.c:
* ext/vorbis/vorbisdec.c:
* ext/vorbis/vorbisenc.c:
* gst-libs/gst/audio/gstaudiofilter.c:
(gst_audio_filter_base_init):
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
(gst_audio_filter_template_base_init):
* gst/adder/gstadder.c: (gst_adder_get_type):
* gst/adder/gstadder.h:
* gst/audioconvert/gstaudioconvert.c:
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init),
(gst_audio_test_src_create):
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/playback/gstdecodebin.c:
* gst/playback/gstplaybin.c:
* gst/playback/gststreamselector.c:
(gst_stream_selector_base_init):
* gst/subparse/gstsubparse.c: (gst_sub_parse_base_init):
* gst/volume/gstvolume.c:
* sys/v4l/gstv4lmjpegsink.c:
* sys/v4l/gstv4lmjpegsrc.c:
* tests/check/libs/cddabasesrc.c:
* tests/old/examples/gob/gst-identity2.gob:
Add docs for adder, use GST_ELEMENT_DETAILS macro,
define GstElementDetails at the top
2006-03-24 10:42:11 +00:00
Wim Taymans
4df07064b8 gst-libs/gst/audio/gstbaseaudiosrc.c: Fix audio sources, forgot to make the ringbuffer startable...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_get_time), (gst_base_audio_src_create),
(gst_base_audio_src_change_state):
Fix audio sources, forgot to make the ringbuffer
startable...
2006-03-23 16:58:03 +00:00
Wim Taymans
2df1088b3f gst-libs/gst/audio/gstbaseaudiosrc.c: unparent instead of unref the ringbuffer.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_get_time), (gst_base_audio_src_create),
(gst_base_audio_src_change_state):
unparent instead of unref the ringbuffer.
2006-03-23 16:29:58 +00:00
Wim Taymans
227474e464 gst-libs/gst/audio/gstbaseaudiosink.c: Implement new async_play vmethod to start slaving and allow playback start in ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init), (gst_base_audio_sink_async_play),
(gst_base_audio_sink_do_play), (gst_base_audio_sink_change_state):
Implement new async_play vmethod to start slaving and allow
playback start in case of async PLAY state changes.
* gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
Enable QoS with new method in base class.
2006-03-23 16:24:23 +00:00
Fabrizio Gennari
1f9a8224f4 gst/: Patch for support of YVU9 AVI files (#334822)
Original commit message from CVS:
Patch by: Fabrizio Gennari <fabrizio dot ge at tiscali dot it>
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
(gst_riff_create_video_template_caps):
* gst/ffmpegcolorspace/avcodec.h:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
(gst_ffmpegcsp_avpicture_fill):
* gst/ffmpegcolorspace/imgconvert.c:
Patch for support of YVU9 AVI files (#334822)
2006-03-23 09:38:59 +00:00
Wim Taymans
747d560fb5 gst-libs/gst/audio/gstbaseaudiosink.c: Since we _parent the ringbuffer, we also need to _unparent instead of a plain ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_dispose):
Since we _parent the ringbuffer, we also need to
_unparent instead of a plain _unref.
2006-03-22 12:33:09 +00:00
Wim Taymans
82fd38fbcf gst-libs/gst/audio/: Only start playback if we are playing. should fix #330748.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_change_state):
* gst-libs/gst/audio/gstringbuffer.c: (wait_segment),
(gst_ring_buffer_may_start):
* gst-libs/gst/audio/gstringbuffer.h:
Only start playback if we are playing.
should fix #330748.
2006-03-17 17:48:33 +00:00
Tim-Philipp Müller
7a683ff848 gst-libs/gst/tag/gstid3tag.c: Add TXXX frame identifiers for replaygain stuff as used by some taggers (see #323721).
Original commit message from CVS:
* gst-libs/gst/tag/gstid3tag.c:
Add TXXX frame identifiers for replaygain stuff as used
by some taggers (see #323721).
2006-03-16 13:08:01 +00:00
Tim-Philipp Müller
c6e2d1eeea gst-libs/gst/tag/gstid3tag.c: Don't crash on unknown ID3v2 TXXX frames.
Original commit message from CVS:
* gst-libs/gst/tag/gstid3tag.c:
Don't crash on unknown ID3v2 TXXX frames.
2006-03-15 22:40:08 +00:00
Jan Schmidt
28bf221806 ext/alsa/gstalsasink.c: Chain up to the parent finalize method.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_finalise):
Chain up to the parent finalize method.
Add 32-bit sample size to the template caps.

* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
(gst_riff_create_video_template_caps):
Add the fourcc that the VMWare codec uses.

* gst/playback/gststreamselector.c:
(gst_stream_selector_set_property),
(gst_stream_selector_bufferalloc),
(gst_stream_selector_request_new_pad):
For the active pad, forward buffer-alloc requests, otherwise
return GST_FLOW_NOT_LINKED. This also prevents xvimagesink
having to memcpy every frame when used by playbin.

* gst/tcp/gstmultifdsink.c:
(gst_multi_fd_sink_handle_client_write):
Get negotiated caps from the sink pad, rather than the sink
pad's peer.
2006-03-15 17:59:05 +00:00
Fabrizio Gennari
36fa065dc9 gst-libs/gst/riff/riff-media.c: Make sure the buffer we copy into is really always big enough, this time for real (#3...
Original commit message from CVS:
Patch by: Fabrizio Gennari  <fabrizio dot ge at tiscali dot it>
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
Make sure the buffer we copy into is really always big
enough, this time for real (#333488).
2006-03-14 15:13:04 +00:00
Tim-Philipp Müller
442722beba gst-libs/gst/riff/riff-media.c: Add support for 24bpp DIB (#305279).
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
Add support for 24bpp DIB (#305279).
2006-03-14 13:16:49 +00:00
Wim Taymans
3150ac3b64 gst/: Re-enable QoS after the release.
Original commit message from CVS:
* gst-libs/gst/video/gstvideofilter.c: (gst_video_filter_init):
* gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
* gst/ffmpegcolorspace/gstffmpegcolorspace.c: (gst_ffmpegcsp_init):
* gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init),
(gst_video_scale_init), (gst_video_scale_src_event):
Re-enable QoS after the release.
Rework videoscale to use the base class src_event handler.
2006-03-14 11:11:59 +00:00
Tim-Philipp Müller
c4ae62d634 docs/plugins/: Add cdparanoiasrc to docs.
Original commit message from CVS:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
Add cdparanoiasrc to docs.
* gst-libs/gst/cdda/gstcddabasesrc.c:
More GstCddaBaseSrc docs.
2006-03-12 14:56:31 +00:00
Tim-Philipp Müller
7762b252a3 Add new API to libgsttag: gst_tag_from_id3_user_tag().
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/tag/gstid3tag.c: (gst_tag_from_id3_user_tag):
* gst-libs/gst/tag/tag.h:
Add new API to libgsttag: gst_tag_from_id3_user_tag().
2006-03-12 13:47:22 +00:00
Jan Schmidt
a9f2d28efb gst-libs/gst/video/gstvideosink.c: Disable max-lateness by setting it to -1 for now, so that we can bed QoS stuff in ...
Original commit message from CVS:
* gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
Disable max-lateness by setting it to -1 for now, so that
we can bed QoS stuff in thoroughly between now and the next
release.
2006-03-10 12:37:53 +00:00
Fabrizio
8d9e3abb5b gst-libs/gst/riff/riff-media.c: Make sure we don't read beyond the palette buffer in case of
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
Make sure we don't read beyond the palette buffer in case of
broken or manipulated files (#333488, patch by: Fabrizio
Gennari)
2006-03-10 11:09:23 +00:00
Wim Taymans
6b9594ac99 ext/libvisual/visual.c: Cleanups, post nice errors.
Original commit message from CVS:
* ext/libvisual/visual.c: (gst_visual_init),
(gst_visual_clear_actors), (gst_visual_dispose),
(gst_visual_reset), (gst_visual_src_setcaps),
(gst_visual_sink_setcaps), (gst_vis_src_negotiate),
(gst_visual_sink_event), (gst_visual_src_event), (get_buffer),
(gst_visual_chain), (gst_visual_change_state):
Cleanups, post nice errors.
Handle sink and src events.
Implement simple QoS.
* gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
Use new basesink methods to configure max-lateness.
Small doc update.
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_transform_caps), (gst_ffmpegcsp_set_caps):
Debug statement cleanups.
* gst/volume/gstvolume.c: (gst_volume_class_init):
Simple cleanup.
2006-03-08 09:53:31 +00:00
Fabrizio
c12d7735ac gst-libs/gst/riff/riff-media.c: Allow palettes with less than 256 colours in AVI files
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
Allow palettes with less than 256 colours in AVI files
(#333488, patch by: Fabrizio Gennari).
2006-03-08 09:37:12 +00:00
Tim-Philipp Müller
ab6f99ab60 gst-libs/gst/audio/gstaudiofilter.c: Don't ignore flow return from gst_pad_push().
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_chain):
Don't ignore flow return from gst_pad_push().
2006-03-07 13:01:21 +00:00
Christophe Fergeau
8e6d3a5c03 Don't leak references returned by gst_pad_get_parent()
Original commit message from CVS:
* ext/libvisual/visual.c: (gst_visual_getcaps),
(gst_visual_src_setcaps), (gst_visual_sink_setcaps):
* ext/ogg/gstoggmux.c: (gst_ogg_mux_sinkconnect):
* ext/vorbis/vorbisenc.c: (gst_vorbisenc_convert_src),
(gst_vorbisenc_convert_sink):
* gst-libs/gst/audio/audio.c: (gst_audio_frame_byte_size),
(gst_audio_duration_from_pad_buffer):
* gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_link),
(gst_audio_filter_chain):
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_setcaps):
* gst-libs/gst/video/video.c: (gst_video_frame_rate),
(gst_video_get_size):
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_setcaps):
Don't leak references returned by gst_pad_get_parent()
(#333663, based on patch by: Christophe Fergeau).
2006-03-07 12:49:03 +00:00
Wim Taymans
a5ff88df2a gst-libs/gst/video/gstvideosink.c: Throw away frames that are later than 20 ms.
Original commit message from CVS:
* gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init),
(gst_video_sink_class_init):
Throw away frames that are later than 20 ms.
2006-03-06 16:18:51 +00:00
Fabrizio
e3abdc2b9a gst-libs/gst/riff/riff-media.c:
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Set depth on WMA caps (#333545, patch by: Fabrizio Gennari).
2006-03-06 14:14:47 +00:00
Tim-Philipp Müller
3fa6e7accd gst-libs/gst/riff/riff-media.c: Add support for '3IVD' fourcc (#333403).
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
Add support for '3IVD' fourcc (#333403).
2006-03-05 13:08:37 +00:00
Wim Taymans
1e9f5c43ad docs/: Added some more docs to libs and plugins.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* docs/libs/gst-plugins-base-libs.types:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
Added some more docs to libs and plugins.
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_prepare_read), (gst_ring_buffer_clear):
* gst-libs/gst/audio/gstringbuffer.h:
Document ringbuffer some more.
* gst/videorate/gstvideorate.c: (gst_video_rate_class_init),
(gst_video_rate_setcaps), (gst_video_rate_reset),
(gst_video_rate_init), (gst_video_rate_flush_prev),
(gst_video_rate_swap_prev), (gst_video_rate_event),
(gst_video_rate_chain), (gst_video_rate_change_state):
* gst/videorate/gstvideorate.h:
Fix videorate to use segments.
Make it work with 0/1 framerates (closes #331903)
Handle EOS correctly.
Added docs.
2006-03-02 16:47:34 +00:00
Wim Taymans
77ff8c9fdb gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to provide a clock in the NULL state.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_provide_clock):
Don't try to provide a clock in the NULL state.
2006-02-28 11:06:24 +00:00
Julien Moutte
0d0354f365 gst-libs/gst/tag/tag.h: Adding Annodex tags here.
Original commit message from CVS:
2006-02-25  Julien MOUTTE  <julien@moutte.net>

* gst-libs/gst/tag/tag.h: Adding Annodex tags here.
2006-02-24 23:19:44 +00:00
Tim-Philipp Müller
7647cafafc gst-libs/gst/riff/riff-media.c:
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
Pick up palette for MS video v1 (#327028, patch by:
Fabrizio Gennari <fabrizio dot get at tiscali dot it>)
2006-02-24 16:21:34 +00:00
Tim-Philipp Müller
043c6d91df gst-libs/gst/audio/mixerutils.c: Make order in which elements are tried more determinable.
Original commit message from CVS:
* gst-libs/gst/audio/mixerutils.c:
(element_factory_rank_compare_func):
Make order in which elements are tried more determinable.
2006-02-20 16:21:14 +00:00
Wim Taymans
3451a81879 gst-libs/gst/audio/gstringbuffer.c: Small cleanups.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_get_type),
(gst_ring_buffer_open_device), (gst_ring_buffer_close_device),
(gst_ring_buffer_device_is_open), (gst_ring_buffer_acquire),
(gst_ring_buffer_release), (gst_ring_buffer_set_flushing),
(gst_ring_buffer_start), (gst_ring_buffer_pause_unlocked),
(gst_ring_buffer_pause), (gst_ring_buffer_stop),
(gst_ring_buffer_delay), (gst_ring_buffer_samples_done),
(gst_ring_buffer_set_sample), (gst_ring_buffer_clear_all),
(gst_ring_buffer_commit), (gst_ring_buffer_prepare_read),
(gst_ring_buffer_clear):
Small cleanups.
Added some G_LIKELY.
2006-02-17 14:07:01 +00:00
Wim Taymans
454618e9b9 gst-libs/gst/audio/TODO: Update TODO
Original commit message from CVS:
* gst-libs/gst/audio/TODO:
Update TODO

* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_get_offset):
When trying to play samples ASAP and we don't have a
previous sample, try to play at position 0 instead of
an invalid position.
2006-02-17 10:15:52 +00:00
Tim-Philipp Müller
8914180cb7 gst-libs/gst/riff/riff-media.c: Set codec_name for WAVEFORMATEX as well. When we have 'normal PCM' with 4 or 6 channe...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Set codec_name for WAVEFORMATEX as well. When we have 'normal PCM'
with 4 or 6 channels, assume a default channel layout to make things
work (not sure there's anything else we can do in those cases).
2006-02-16 20:19:51 +00:00
Tim-Philipp Müller
9490d413c0 gst-libs/gst/audio/multichannel.c: Minor docs fix.
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
Minor docs fix.
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/riff/riff-ids.h:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_wavext_add_channel_layout), (gst_riff_create_audio_caps):
Add support for WAVEFORMATEX, eg. PCM audio with more than two
channels and a channel layout map.
2006-02-16 19:18:46 +00:00
Tim-Philipp Müller
5b788a8a66 gst-libs/gst/audio/multichannel.c: When we have more than 2 channels, but no channel layout is specified in the caps,...
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_get_channel_positions):
When we have more than 2 channels, but no channel layout is
specified in the caps, return some default channel layout
to the caller and warn about about a possibly buggy element
(could be buggy filtercaps as well of course) (#317038).
2006-02-16 11:44:43 +00:00
Wim Taymans
3b45740289 gst-libs/gst/audio/gstringbuffer.c: Add some compiler G_(UN_)LIKELY help.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_pause_unlocked), (gst_ring_buffer_delay),
(gst_ring_buffer_samples_done), (wait_segment),
(gst_ring_buffer_commit), (gst_ring_buffer_clear):
Add some compiler G_(UN_)LIKELY help.
SIGNAL the ringbuffer waiters when going to PAUSED as well to
make sure they can exit their functions. Should fix #330748
2006-02-14 13:45:35 +00:00
Wim Taymans
16dbdc5c21 gst-libs/gst/audio/gstbaseaudiosink.c: Always sync on first sample we receive when starting.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_get_time), (gst_base_audio_sink_preroll),
(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
Always sync on first sample we receive when starting.
2006-02-13 18:49:02 +00:00
Tim-Philipp Müller
c60b2b46b2 gst-libs/gst/cdda/gstcddabasesrc.c: No need to push an EOS event here, GstBaseSrc will do that for us when we return ...
Original commit message from CVS:
* gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
No need to push an EOS event here, GstBaseSrc will do that for us
when we return FLOW_UNEXPECTED.
2006-02-13 15:17:34 +00:00
Wim Taymans
0be7d56eb9 gst-libs/gst/audio/gstbaseaudiosink.c: Use scale functions when possible.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_get_time), (gst_base_audio_sink_setcaps),
(gst_base_audio_sink_drain), (gst_base_audio_sink_preroll),
(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
Use scale functions when possible.
Fix error messages.
Free clockid when after waiting for EOS.
Use G_(UN_)LIKLY when it makes sense.
Fix sample clipping bug found by Arwed v. Merkatz fixes #330789.
2006-02-12 14:54:55 +00:00
Edward Hervey
b6d4e180ce gst-libs/gst/riff/riff-media.c: Added FLV1 <==> 'video/x-flash-video,flvversion=1' conversion.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
(gst_riff_create_video_template_caps):
Added FLV1 <==> 'video/x-flash-video,flvversion=1' conversion.
Also added the caps to the default set of riff video caps.
2006-02-10 10:53:33 +00:00
Thomas Vander Stichele
fdaa7a7a00 gst-libs/gst/rtp/gstbasertppayload.c: update seqnum before setting it on the packet; this makes sure that the timesta...
Original commit message from CVS:

* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_setcaps), (gst_basertppayload_push):
update seqnum before setting it on the packet; this makes sure
that the timestamp and seqnum properties match after pushing
a buffer
2006-02-09 17:04:18 +00:00
Andy Wingo
4e0c846fa4 kapowpowpow
Original commit message from CVS:
kapowpowpow
2006-02-09 11:46:03 +00:00
Andy Wingo
4ae63e7361 gst-libs/gst/audio/gstringbuffer.c
Original commit message from CVS:
2006-02-09  Andy Wingo  <wingo@pobox.com>

* gst-libs/gst/audio/gstringbuffer.c
(gst_ring_buffer_samples_done): Cast to guint64, fixes an integer
overflow after 13.5 hours of recording. Kapow!

* ext/alsa/gstalsasrc.c (gst_alsasrc_delay): Clamp the delay to
the buffer size -- we don't care about underrun/overrun reporting
right now, just need to return a useful value.
2006-02-09 11:36:18 +00:00
Tim-Philipp Müller
148b83dd20 gst-libs/gst/tag/gstvorbistag.c: Make sure we called gst_tag_register_musicbrainz_tags() before possibly mapping a vo...
Original commit message from CVS:
* gst-libs/gst/tag/gstvorbistag.c: (gst_tag_from_vorbis_tag),
(gst_tag_to_vorbis_tag):
Make sure we called gst_tag_register_musicbrainz_tags()
before possibly mapping a vorbiscomment string from/to a
musicbrainz tag.
2006-02-07 17:44:53 +00:00
Jan Schmidt
a81e97c96e gst-libs/gst/tag/gstid3tag.c: Add mapping for ID3 International Standard Recording Code tag "TSRC"
Original commit message from CVS:
* gst-libs/gst/tag/gstid3tag.c:
Add mapping for ID3 International Standard Recording Code
tag "TSRC"
2006-02-05 22:47:41 +00:00
Tim-Philipp Müller
7b43847c2c Split libgsttag docs into multiple sections.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/tag/gstid3tag.c:
* gst-libs/gst/tag/gstvorbistag.c:
* gst-libs/gst/tag/tags.c:
Split libgsttag docs into multiple sections.
2006-02-05 18:22:01 +00:00
Tim-Philipp Müller
8db163abcf Add libgsttag to the docs.
Original commit message from CVS:
* docs/libs/Makefile.am:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/tag/gstid3tag.c: (gst_tag_from_id3_tag):
* gst-libs/gst/tag/gstvorbistag.c:
* gst-libs/gst/tag/tag.h:
* gst-libs/gst/tag/tags.c:
Add libgsttag to the docs.
2006-02-05 18:01:33 +00:00
Tim-Philipp Müller
00e0574b1f gst-libs/gst/tag/gstvorbistag.c: Add comment about LANGUAGE tag inconsistency (we want
Original commit message from CVS:
* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
Add comment about LANGUAGE tag inconsistency (we want
ISO-639-1, but extract three-letter identifiers?)
* po/POTFILES.in:
Add two translatable files.
2006-02-05 15:14:06 +00:00
Tim-Philipp Müller
ce175c9953 gst-libs/gst/tag/: Forward-port some tags stuff from the 0.8 branch. This is mostly the addition of musicbrainz tags ...
Original commit message from CVS:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
* gst-libs/gst/tag/tag.h:
* gst-libs/gst/tag/tags.c:
(gst_tag_register_musicbrainz_tags_internal),
(gst_tag_register_musicbrainz_tags):
Forward-port some tags stuff from the 0.8 branch. This is
mostly the addition of musicbrainz tags and their mapping
to vorbistags, and a vorbistag mapping of the language tag.
2006-02-05 14:59:28 +00:00
Wim Taymans
260b5295c9 gst-libs/gst/audio/gstbaseaudiosink.c: Ugh.. getting late I guess...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_provide_clock):
Ugh.. getting late I guess...
2006-02-02 18:18:31 +00:00
Wim Taymans
c78a5d7e1e gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to provide a clock when we are not negotiated since we might not be ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_provide_clock),
(gst_base_audio_sink_set_property),
(gst_base_audio_sink_get_property), (gst_base_audio_sink_render):
Don't try to provide a clock when we are not negotiated since
we might not be able to make it run.
2006-02-02 18:13:26 +00:00
Wim Taymans
416c011f11 gst-libs/gst/audio/TODO: Updated.
Original commit message from CVS:
* gst-libs/gst/audio/TODO:
Updated.

* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain), (gst_base_audio_sink_event):
On EOS, wait till the last sample is played before posting EOS.
2006-02-02 12:14:35 +00:00
Kai Vehmanen
a6d1e0b5bf gst-libs/gst/rtp/gstbasertpdepayload.c: setting queue_delay to zero. Also avoid thread being started if queue_delay i...
Original commit message from CVS:
2006-02-01  Philippe Kalaf <burger at speedy dot org>

* gst-libs/gst/rtp/gstbasertpdepayload.c:
Patch by Kai Vehmanen : Adds ability to enable newsegment bypass by
setting queue_delay to zero. Also avoid thread being started if
queue_delay is zero.
2006-02-02 00:04:37 +00:00
Tim-Philipp Müller
51ce1f6179 gst/: Pass unhandled queries to the parent class's query function.
Original commit message from CVS:
* gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_query),
(gst_cdda_base_src_handle_event):
* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_query):
Pass unhandled queries to the parent class's query function.
2006-02-01 11:59:47 +00:00
Wim Taymans
a169abc679 gst-libs/gst/audio/gstaudiosink.c: Implement pause that does not wait for completion.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c:
(gst_audioringbuffer_class_init), (gst_audioringbuffer_release),
(gst_audioringbuffer_pause):
Implement pause that does not wait for completion.

* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
Don't drop buffers when going to PAUSED but perform preroll on
remaining samples now that core base class supports this.

* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_release),
(gst_ring_buffer_pause_unlocked), (gst_ring_buffer_stop),
(gst_ring_buffer_commit):
Pause should not signal waiters.
Implement return value of _commit correctly.
2006-01-30 16:19:33 +00:00
Sébastien Moutte
dc46970cdf gst-libs/gst/audio/audio.h: (GST_CLOCK_TIME_TO_FRAMES)
Original commit message from CVS:
* gst-libs/gst/audio/audio.h: (GST_CLOCK_TIME_TO_FRAMES)
* gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_render)
use of gst_guint64_to_gdouble to be compliant with vs6
* gst/playback/gstdecodebin.c: (try_to_link_1)
* gst/videorate/videorate.c: (gst_video_rate_blank_data)
use of G_GINT64_CONSTANT for int64 constants
* win32/common/libgstinterfaces.def:
export some symbols (gst_mixer_get_type,gst_mixer_track_get_type)
* win32/vs6:
update and add new project files
2006-01-29 19:13:39 +00:00
Tim-Philipp Müller
27ed152e10 gst-libs/gst/audio/gstringbuffer.c: Make gcc-4.1 happy (part of #327357).
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (build_linear_format):
Make gcc-4.1 happy (part of #327357).
2006-01-28 18:19:18 +00:00
Wim Taymans
ccd05fa086 gst-libs/gst/audio/gstbaseaudiosink.c: Undo previous commit, it breaks resume after pause.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
Undo previous commit, it breaks resume after pause.
2006-01-25 10:50:32 +00:00
Wim Taymans
2bc5ca1786 gst-libs/gst/audio/gstbaseaudiosink.c: Improve debugging.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_setcaps), (gst_base_audio_sink_event),
(gst_base_audio_sink_preroll), (gst_base_audio_sink_render):
Improve debugging.
Post error when caps cannot be parsed.
Resync on discontinuity in the stream.
Clip samples to segment boundaries.
return WRONG_STATE sooner when we are flushing.

* gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
(gst_base_audio_src_get_time), (gst_base_audio_src_create):
Make audiosrc operate in TIME.
Set TIMESTAMP and DURATION on buffers.
2006-01-25 09:27:01 +00:00
Jens Granseuer
6b153515b0 GCC 2.95 fixes (#328263).
Original commit message from CVS:
2006-01-23  Andy Wingo  <wingo@pobox.com>

* ext/alsa/gstalsasink.c:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_setcaps),
(gst_base_rtp_depayload_add_to_queue),
(gst_base_rtp_depayload_queue_release): GCC 2.95 fixes (#328263).

Patch by: Jens Granseuer <jensgr at gmx dot net>
2006-01-23 10:10:36 +00:00
Tim-Philipp Müller
4757506720 gst-libs/gst/interfaces/xoverlay.c: Fix prepare-xwindow-id code example in the docs - we need to ignore all messages ...
Original commit message from CVS:
* gst-libs/gst/interfaces/xoverlay.c:
Fix prepare-xwindow-id code example in the docs - we need to
ignore all messages that aren't element messages as well.
2006-01-22 14:50:53 +00:00
Jan Schmidt
04333a568c gst-libs/gst/audio/gstbaseaudiosink.c: Fix playback of non-synchronised streams by assuming a rate of 1.0 instead of ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
Fix playback of non-synchronised streams by assuming a rate
of 1.0 instead of a random one.

Makes this work again:

gst-launch filesrc location=raw_audio.file ! 'audio/x-raw-int,
endianness=(int)4321, signed=(boolean)true, width=(int)16,
depth=(int)16, rate=(int)44100, channels=(int)2' ! audioconvert !
audioresample ! alsasink
2006-01-17 11:43:49 +00:00
Tim-Philipp Müller
f220f8295b Add docs for mixerutils stuff.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/mixerutils.c:
* gst-libs/gst/audio/mixerutils.h:
Add docs for mixerutils stuff.
2006-01-14 12:52:22 +00:00
Thomas Vander Stichele
5fd8ee2ea4 gst-libs/gst/audio/mixerutils.c: actually save the element we create
Original commit message from CVS:

* gst-libs/gst/audio/mixerutils.c:
(gst_audio_mixer_filter_do_filter):
actually save the element we create
2006-01-13 16:45:50 +00:00
Tim-Philipp Müller
c0b087e4b7 gst-libs/gst/cdda/gstcddabasesrc.c: No need to post a tag message on the bus when seeking within the same track, only...
Original commit message from CVS:
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_handle_track_seek):
No need to post a tag message on the bus when seeking
within the same track, only post it when the current
track changes.
2006-01-12 14:56:11 +00:00
Tim-Philipp Müller
b867510721 gst-libs/gst/audio/gstringbuffer.c: Set depth and width for alaw/mulaw (fixes #326601).
Original commit message from CVS:
Reviewed by: Tim-Philipp Müller  <tim at centricular dot net>
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps):
Set depth and width for alaw/mulaw (fixes #326601).
2006-01-11 15:11:20 +00:00
Michael Smith
b0c21cab17 gst-libs/gst/audio/gstaudiosrc.c: Don't leak GCond in audio sources.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_dispose):
Don't leak GCond in audio sources.
2006-01-10 12:25:59 +00:00
Tim-Philipp Müller
8ec22e812b gst-libs/gst/audio/gstringbuffer.*: Name (private) union, makes Forte compiler happy (this time for real) (#324900).
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_init),
(gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
(gst_ring_buffer_pause), (wait_segment):
* gst-libs/gst/audio/gstringbuffer.h:
Name (private) union, makes Forte compiler happy (this time
for real) (#324900).
2006-01-10 09:38:44 +00:00
Tim-Philipp Müller
3b96467f63 gst-libs/gst/audio/Makefile.am: Link against libgstinterfaces, needed for mixer and property probe stuff.
Original commit message from CVS:
* gst-libs/gst/audio/Makefile.am:
Link against libgstinterfaces, needed for mixer
and property probe stuff.
2006-01-09 10:52:33 +00:00
Edward Hervey
8a273afec0 gst-libs/gst/Makefile.am:
Original commit message from CVS:
* gst-libs/gst/Makefile.am:
2006-01-09 10:46:52 +00:00
Tim-Philipp Müller
e737f441d3 gst-libs/gst/audio/: Add gst_audio_default_registry_mixer_filter() utility function.
Original commit message from CVS:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/audio/mixerutils.c:
(gst_audio_mixer_filter_do_filter),
(gst_audio_mixer_filter_check_element),
(gst_audio_mixer_filter_probe_feature),
(element_factory_rank_compare_func),
(gst_audio_default_registry_mixer_filter):
* gst-libs/gst/audio/mixerutils.h:
Add gst_audio_default_registry_mixer_filter() utility
function.
2006-01-09 09:38:34 +00:00
Tim-Philipp Müller
be8f055317 gst-libs/gst/audio/gstringbuffer.*: Sun's Forte compiler doesn't seem to like anonymous structs, so use same setup as...
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_init),
(gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
(gst_ring_buffer_pause), (wait_segment):
* gst-libs/gst/audio/gstringbuffer.h:
Sun's Forte compiler doesn't seem to like anonymous structs,
so use same setup as in GstBaseSrc (fixes #324900).
2006-01-02 23:37:38 +00:00
Tim-Philipp Müller
1068588993 gst-libs/gst/cdda/gstcddabasesrc.c: An integer is not a string. Fix access to uninitialised variable.
Original commit message from CVS:
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_update_duration),
(gst_cdda_base_src_calculate_cddb_id):
An integer is not a string. Fix access to uninitialised variable.
* tests/check/Makefile.am:
Add cddabasesrc unit test; also actually enable the vorbis test.
* tests/check/generic/states.c:
Blacklist new cd audio elements as well.
* tests/check/libs/cddabasesrc.c:
Unit test for GstCddaBaseSrc (discid calculation mostly).
2005-12-29 11:49:11 +00:00
Tim-Philipp Müller
27af2b5ddc docs/libs/: Add docs for libgstcdda/GstCddaBaseSrc.
Original commit message from CVS:
* docs/libs/Makefile.am:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* docs/libs/gst-plugins-base-libs.types:
Add docs for libgstcdda/GstCddaBaseSrc.
* gst-libs/gst/interfaces/mixertrack.h:
Do one struct member per line with a semicolon at the end, that way
even gtk-doc might parse it without complaining.
2005-12-28 18:19:25 +00:00
Tim-Philipp Müller
d129bea2be Add new libgstcdda with GstCddaBaseSrc class.
Original commit message from CVS:
* configure.ac:
* gst-libs/gst/Makefile.am:
* gst-libs/gst/cdda/Makefile.am:
* gst-libs/gst/cdda/base64.c:
* gst-libs/gst/cdda/base64.h:
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_mode_get_type), (gst_cdda_base_src_base_init),
(gst_cdda_base_src_class_init), (gst_cdda_base_src_init),
(gst_cdda_base_src_finalize), (gst_cdda_base_src_set_property),
(gst_cdda_base_src_get_property),
(gst_cdda_base_src_get_track_from_sector),
(gst_cdda_base_src_get_query_types), (gst_cdda_base_src_convert),
(gst_cdda_base_src_query), (gst_cdda_base_src_is_seekable),
(gst_cdda_base_src_do_seek), (gst_cdda_base_src_handle_track_seek),
(gst_cdda_base_src_handle_event), (gst_cdda_base_src_uri_get_type),
(gst_cdda_base_src_uri_get_protocols),
(gst_cdda_base_src_uri_get_uri), (gst_cdda_base_src_uri_set_uri),
(gst_cdda_base_src_uri_handler_init),
(gst_cdda_base_src_setup_interfaces),
(gst_cdda_base_src_add_track), (gst_cdda_base_src_update_duration),
(cddb_sum), (gst_cddabasesrc_calculate_musicbrainz_discid),
(lba_to_msf), (gst_cdda_base_src_calculate_cddb_id),
(gst_cdda_base_src_add_tags),
(gst_cdda_base_src_add_index_associations),
(gst_cdda_base_src_set_index), (gst_cdda_base_src_get_index),
(gst_cdda_base_src_track_sort_func), (gst_cdda_base_src_start),
(gst_cdda_base_src_clear_tracks), (gst_cdda_base_src_stop),
(gst_cdda_base_src_create):
* gst-libs/gst/cdda/gstcddabasesrc.h:
* gst-libs/gst/cdda/sha1.c:
* gst-libs/gst/cdda/sha1.h:
Add new libgstcdda with GstCddaBaseSrc class.
2005-12-28 18:06:50 +00:00
Thomas Vander Stichele
01bc68f918 stop making fun of older compilers
Original commit message from CVS:
stop making fun of older compilers
2005-12-20 12:24:29 +00:00
Thomas Vander Stichele
b4b2b62a74 gst-libs/gst/audio/: update strings, values are in microseconds change the default sink buffer time to something that...
Original commit message from CVS:

* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init):
update strings, values are in microseconds
change the default sink buffer time to something that is smaller
(to help software volume mixing have a slightly lower delay) but
still be acceptable on Wim's laptop
2005-12-20 12:00:26 +00:00
Edward Hervey
ffc33e35d9 gst-libs/gst/riff/riff-media.c: Made a quack, forgot to add DUCK to the riff video template.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_template_caps):
Made a quack, forgot to add DUCK to the riff video template.
2005-12-20 10:13:05 +00:00
Edward Hervey
e54ffc2c5f ext/ogg/gstogmparse.c: Make sure pads are initialized correctly.
Original commit message from CVS:
* ext/ogg/gstogmparse.c: (gst_ogm_text_parse_base_init),
(gst_ogm_parse_init), (gst_ogm_audio_parse_init),
(gst_ogm_video_parse_init), (gst_ogm_text_parse_init),
(gst_ogm_parse_chain):
Make sure pads are initialized correctly.
* gst-libs/gst/riff/riff-ids.h:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
(gst_riff_create_video_template_caps):
Add a whole bunch of FOURCC <=> MimeType.
Extend the riff video pad template to support the newly added fourcc.
2005-12-19 15:00:38 +00:00
Julien Moutte
7811799ce8 gst-libs/gst/video/gstvideosink.c: Add a debug category.
Original commit message from CVS:
2005-12-18  Julien MOUTTE  <julien@moutte.net>

* gst-libs/gst/video/gstvideosink.c: (gst_video_sink_base_init),
(gst_video_sink_get_type): Add a debug category.
2005-12-18 10:54:33 +00:00
Philippe Kalaf
f5723a256e gst-libs/gst/rtp/gstbasertpdepayload.c: Handle downstream newsegment by sending our own newsegment before the next bu...
Original commit message from CVS:
2005-12-17  Philippe Khalaf  <burger@speedy.org>

* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_init), (gst_base_rtp_depayload_handle_sink_event):
Handle downstream newsegment by sending our own newsegment before the
next buffer to be released. (#323900)
2005-12-18 00:56:07 +00:00
Philippe Kalaf
3247ffbbb3 gst-libs/gst/rtp/gstbasertpdepayload.c: add queue delay to new segment as well (as opposed to just the first buffer)....
Original commit message from CVS:
2005-12-17  Philippe Khalaf  <burger@speedy.org>

* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_set_gst_timestamp):
add queue delay to new segment as well (as opposed to just the first
buffer). (bug #322347)
2005-12-18 00:41:10 +00:00
Thomas Vander Stichele
5f83aa7dfa expand tabs
Original commit message from CVS:
expand tabs
2005-12-06 19:42:02 +00:00
Kai Vehmanen
b515b87da8 gst-libs/gst/rtp/gstbasertpdepayload.c
Original commit message from CVS:
2005-12-05  Andy Wingo  <wingo@pobox.com>

patch by: Kai Vehmanen <kv2004 eca cx>

* gst-libs/gst/rtp/gstbasertpdepayload.c
(gst_base_rtp_depayload_thread): Fix busy loop (#323017).
2005-12-05 10:47:55 +00:00
Thomas Vander Stichele
ff18d8e298 Do burger's rename for rtp payloaders and depayloaders
Original commit message from CVS:

* docs/libs/tmpl/gstcolorbalance.sgml:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
* gst-libs/gst/rtp/gstbasertppayload.c:
* gst-libs/gst/rtp/gstrtpbuffer.c:
* gst-libs/gst/rtp/gstrtpbuffer.h:
Do burger's rename for rtp payloaders and depayloaders
2005-12-01 14:29:59 +00:00
Thomas Vander Stichele
9db2e7681a borgify
Original commit message from CVS:
borgify
2005-12-01 14:29:09 +00:00
Thomas Vander Stichele
dc0f45d20c update documentation
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* docs/libs/tmpl/gstaudio.sgml:
* docs/libs/tmpl/gstringbuffer.sgml:
* gst-libs/gst/interfaces/xoverlay.c:
* gst-libs/gst/video/gstvideofilter.c:
* gst-libs/gst/video/gstvideosink.c:
update documentation
2005-12-01 01:29:46 +00:00
Wim Taymans
109cd71c76 gst-libs/gst/rtp/: closed #320644.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.h:
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
(gst_basertppayload_event), (gst_basertppayload_push),
(gst_basertppayload_change_state):
* gst-libs/gst/rtp/gstbasertppayload.h:
closed #320644.
2005-11-30 19:08:14 +00:00
Julien Moutte
ef53753a6b Adding docs.
Original commit message from CVS:
2005-11-30  Julien MOUTTE  <julien@moutte.net>

* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/video/gstvideofilter.c:
* gst-libs/gst/video/gstvideosink.c:
* gst-libs/gst/video/gstvideosink.h: Adding docs.
2005-11-30 18:57:48 +00:00
Thomas Vander Stichele
2a1a6b755b gst-libs/gst/net/: this was moved to netbuffer
Original commit message from CVS:

* gst-libs/gst/net/Makefile.am:
* gst-libs/gst/net/README:
* gst-libs/gst/net/gstnetbuffer.c:
* gst-libs/gst/net/gstnetbuffer.h:
this was moved to netbuffer
2005-11-30 10:40:09 +00:00
Thomas Vander Stichele
d50aed7960 gst-libs/gst/video/gstvideofilter.*: borgify name to bring in line with other classes
Original commit message from CVS:

* gst-libs/gst/video/gstvideofilter.c: (gst_video_filter_get_type),
(gst_video_filter_class_init), (gst_video_filter_init):
* gst-libs/gst/video/gstvideofilter.h:
borgify name to bring in line with other classes
2005-11-30 08:26:39 +00:00
Edward Hervey
573b0b8166 gst-libs/gst/netbuffer/Makefile.am: really, really tired
Original commit message from CVS:
* gst-libs/gst/netbuffer/Makefile.am:
really, really tired
2005-11-30 00:11:51 +00:00
Edward Hervey
41e620af9e gst-libs/gst/netbuffer/Makefile.am: Let's not override libgstnet from core for no reason...
Original commit message from CVS:
* gst-libs/gst/netbuffer/Makefile.am: (libgstnetbufferincludedir):
Let's not override libgstnet from core for no reason...
(libgstnetbuffer_@GST_MAJORMINOR@_la_SOURCES):
Ok, maybe not so quick next time.
2005-11-29 23:54:37 +00:00
Edward Hervey
a54de3a653 gst-libs/gst/netbuffer/Makefile.am: Let's not override libgstnet from core for no reason...
Original commit message from CVS:
* gst-libs/gst/netbuffer/Makefile.am: (libgstnetbufferincludedir):
Let's not override libgstnet from core for no reason...
2005-11-29 23:51:28 +00:00
Thomas Vander Stichele
2f539d3fa1 fix install location of netbuffer
Original commit message from CVS:
fix install location of netbuffer
2005-11-29 19:16:15 +00:00
Thomas Vander Stichele
31a2b5b408 moved gst-libs/gst/net to netbuffer through CVS surgery remove old directory updating build to accomodate (#322257)
Original commit message from CVS:

* configure.ac:
* gst-libs/gst/Makefile.am:
moved gst-libs/gst/net to netbuffer through CVS surgery
remove old directory
updating build to accomodate
(#322257)
2005-11-29 17:46:37 +00:00
Andy Wingo
dfdde20c6c Rename gstnet to gstnetbuffer (#322257).
Original commit message from CVS:
2005-11-29  Andy Wingo  <wingo@pobox.com>

* pkgconfig/gstreamer-plugins-base.pc.in:
* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
* gst-libs/gst/net/Makefile.am: Rename gstnet to gstnetbuffer
(#322257).
2005-11-29 17:41:16 +00:00
Thomas Vander Stichele
8823933bcd folded audiofilter into the audio library
Original commit message from CVS:
folded audiofilter into the audio library
2005-11-29 01:25:31 +00:00
Thomas Vander Stichele
9c8815566e include the right header
Original commit message from CVS:
include the right header
2005-11-28 17:42:57 +00:00
Thomas Vander Stichele
4d8c9991a4 gst-libs/gst/video/Makefile.am: compile in copied-over videofilter into the video library
Original commit message from CVS:

* gst-libs/gst/video/Makefile.am:
compile in copied-over videofilter into the video library
* gst-libs/gst/video/videosink.h:
rename the header to gstvideosink.h since it's a base GstObject class
* sys/ximage/ximagesink.h:
* sys/xvimage/xvimagesink.h:
use the new header
2005-11-28 17:32:55 +00:00
Wim Taymans
3f05db1828 gst-libs/gst/audio/TODO: Updated TODO
Original commit message from CVS:
* gst-libs/gst/audio/TODO:
Updated TODO

* gst-libs/gst/audio/gstaudiosink.c:
(gst_audioringbuffer_open_device),
(gst_audioringbuffer_close_device), (gst_audioringbuffer_acquire),
(gst_audioringbuffer_release):
Small cleanups.

* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init), (gst_base_audio_sink_render),
(gst_base_audio_sink_change_state):
Slave to the master clock when going to PLAYING and unslave when
going to PAUSED.

* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_open_device), (gst_ring_buffer_close_device),
(gst_ring_buffer_acquire), (gst_ring_buffer_release),
(gst_ring_buffer_samples_done), (gst_ring_buffer_set_sample),
(gst_ring_buffer_clear_all), (wait_segment),
(gst_ring_buffer_commit), (gst_ring_buffer_read),
(gst_ring_buffer_advance):
* gst-libs/gst/audio/gstringbuffer.h:
Add some docs and cleanups.
2005-11-28 15:53:55 +00:00
Julien Moutte
4a3356cd6f Add docs for the Xoverlay interface.
Original commit message from CVS:
2005-11-28  Julien MOUTTE  <julien@moutte.net>

* docs/libs/tmpl/gstcolorbalance.sgml:
* docs/libs/tmpl/gstmixer.sgml:
* docs/libs/tmpl/gstxoverlay.sgml:
* gst-libs/gst/interfaces/xoverlay.c: Add docs for the Xoverlay
interface.
2005-11-28 13:06:05 +00:00
Edward Hervey
b80fd557a2 gst-libs/gst/riff/riff-media.c: Add ATRAC3 to the list of riff-possible audio caps.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_audio_template_caps):
Add ATRAC3 to the list of riff-possible audio caps.
I know we still don't have a plugin for atrac3, but it's saner to output
that than a cryptic mimetype.
2005-11-28 10:12:26 +00:00
Edward Hervey
89df316769 gst-libs/gst/riff/riff-read.c: Don't try to create a zero-sized subbuffer.
Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c: (gst_riff_parse_strf_auds):
Don't try to create a zero-sized subbuffer.
2005-11-27 19:18:31 +00:00
Thomas Vander Stichele
1a4ad8377d gst-libs/gst/: remove silly include
Original commit message from CVS:

* gst-libs/gst/net/Makefile.am:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/tag/Makefile.am:
remove silly include
* gst/tags/Makefile.am:
* gst/tags/gsttagediting.c:
* gst/tags/gsttageditingprivate.h:
* gst/tags/tagedit.vcproj:
remove directory, is as good as empty
2005-11-27 16:27:20 +00:00
Thomas Vander Stichele
efb938bd9a configure.ac: added GST_LIB_LDFLAGS and GST_ALL_LDFLAGS
Original commit message from CVS:

* configure.ac:
added GST_LIB_LDFLAGS and GST_ALL_LDFLAGS
* gst-libs/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/net/Makefile.am:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/video/Makefile.am:
and use them
2005-11-27 16:18:50 +00:00
Julien Moutte
65893762cd VideoFilter inherits from
Original commit message from CVS:
2005-11-23  Julien MOUTTE  <julien@moutte.net>

* ext/cairo/gsttimeoverlay.c:
(gst_timeoverlay_update_font_height),
(gst_timeoverlay_set_caps), (gst_timeoverlay_get_unit_size),
(gst_timeoverlay_transform), (gst_timeoverlay_base_init),
(gst_timeoverlay_class_init), (gst_timeoverlay_init),
(gst_timeoverlay_get_type):
* ext/cairo/gsttimeoverlay.h:
* gst/debug/Makefile.am:
* gst/debug/gstnavigationtest.c:
(gst_navigationtest_handle_src_event),
(gst_navigationtest_get_unit_size),
(gst_navigationtest_set_caps),
(gst_navigationtest_transform),
(gst_navigationtest_change_state),
(gst_navigationtest_base_init), (gst_navigationtest_class_init),
(gst_navigationtest_init), (gst_navigationtest_get_type),
(plugin_init):
* gst/debug/gstnavigationtest.h:
* gst/effectv/Makefile.am:
* gst/effectv/gstaging.c: (gst_agingtv_set_caps),
(gst_agingtv_get_unit_size), (gst_agingtv_transform),
(gst_agingtv_base_init), (gst_agingtv_class_init),
(gst_agingtv_init), (gst_agingtv_get_type):
* gst/effectv/gstdice.c: (gst_dicetv_set_caps),
(gst_dicetv_get_unit_size), (gst_dicetv_transform),
(gst_dicetv_base_init), (gst_dicetv_class_init),
(gst_dicetv_init),
(gst_dicetv_get_type):
* gst/effectv/gstedge.c: (gst_edgetv_set_caps),
(gst_edgetv_get_unit_size), (gst_edgetv_transform),
(gst_edgetv_base_init), (gst_edgetv_class_init),
(gst_edgetv_init),
(gst_edgetv_get_type):
* gst/effectv/gsteffectv.c:
* gst/effectv/gsteffectv.h:
* gst/effectv/gstquark.c: (gst_quarktv_set_caps),
(gst_quarktv_get_unit_size), (fastrand),
(gst_quarktv_transform),
(gst_quarktv_change_state), (gst_quarktv_base_init),
(gst_quarktv_class_init), (gst_quarktv_init),
(gst_quarktv_get_type):
* gst/effectv/gstrev.c: (gst_revtv_set_caps),
(gst_revtv_get_unit_size), (gst_revtv_transform),
(gst_revtv_base_init), (gst_revtv_class_init), (gst_revtv_init),
(gst_revtv_get_type):
* gst/effectv/gstshagadelic.c: (gst_shagadelictv_set_caps),
(gst_shagadelictv_get_unit_size), (gst_shagadelictv_transform),
(gst_shagadelictv_base_init), (gst_shagadelictv_class_init),
(gst_shagadelictv_init), (gst_shagadelictv_get_type):
* gst/effectv/gstvertigo.c: (gst_vertigotv_set_caps),
(gst_vertigotv_get_unit_size), (gst_vertigotv_transform),
(gst_vertigotv_base_init), (gst_vertigotv_class_init),
(gst_vertigotv_init), (gst_vertigotv_get_type):
* gst/effectv/gstwarp.c: (gst_warptv_set_caps),
(gst_warptv_get_unit_size), (gst_warptv_transform),
(gst_warptv_base_init), (gst_warptv_class_init),
(gst_warptv_init),
(gst_warptv_get_type):
* gst/videofilter/Makefile.am:
* gst/videofilter/gstvideobalance.c:
* gst/videofilter/gstvideobalance.h:
* gst/videofilter/gstvideofilter.c: (gst_videofilter_get_type),
(gst_videofilter_class_init), (gst_videofilter_init):
* gst/videofilter/gstvideofilter.h:
* gst/videofilter/gstvideoflip.c: (gst_videoflip_set_caps),
(gst_videoflip_transform_caps), (gst_videoflip_get_unit_size),
(gst_videoflip_flip), (gst_videoflip_transform),
(gst_videoflip_handle_src_event), (gst_videoflip_set_property),
(gst_videoflip_base_init), (gst_videoflip_class_init),
(gst_videoflip_init), (plugin_init), (gst_videoflip_get_type):
* gst/videofilter/gstvideoflip.h: VideoFilter inherits from
BaseTransform, it's just a place holder for now and every video
effect plugin has been ported to use BaseTransform features
directly. QuarkTV was fixed too (was broken), navigationtest
works
and best for the end, videoflip converts navigation events
depending
on flip method ! Fixes #320953
2005-11-23 15:50:51 +00:00
Wim Taymans
c7dc33e495 gst-libs/gst/audio/gstringbuffer.c: If we are reading too slowly, jump forward in the ringbuffer instead of blocking.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_read):
If we are reading too slowly, jump forward in the ringbuffer
instead of blocking.
2005-11-23 13:29:50 +00:00
Wim Taymans
67b21a9033 gst-libs/gst/audio/gstbaseaudiosink.c: Fix for calibration API change.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init), (gst_base_audio_sink_set_clock),
(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
Fix for calibration API change.
2005-11-23 13:08:54 +00:00
Michael Smith
71f3969208 gst-libs/gst/audio/multichannel.c: Use gst_value_array_*() functions on value arrays, not gst_value_list_*().
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_get_channel_positions),
(gst_audio_set_channel_positions),
(gst_audio_set_structure_channel_positions_list),
(gst_audio_fixate_channel_positions):
Use gst_value_array_*() functions on value arrays, not
gst_value_list_*().
2005-11-23 12:40:04 +00:00
Jan Schmidt
54ec8cf70e ext/libvisual/visual.c: Fix the fps calculations.
Original commit message from CVS:
* ext/libvisual/visual.c: (gst_visual_chain):
Fix the fps calculations.

* gst/ffmpegcolorspace/avcodec.h:
Move structure element for clarity

* gst-libs/gst/interfaces/tunernorm.c: (gst_tuner_norm_init):
* gst-libs/gst/interfaces/tunernorm.h:
* sys/v4l/gstv4ljpegsrc.c: (gst_v4ljpegsrc_src_link):
* sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_base_init):
* sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_base_init),
(gst_v4lmjpegsrc_get_fps), (gst_v4lmjpegsrc_src_convert),
(gst_v4lmjpegsrc_src_query), (gst_v4lmjpegsrc_get),
(gst_v4lmjpegsrc_getcaps):
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_fixate), (gst_v4lsrc_get_caps),
(gst_v4lsrc_set_caps):
* sys/v4l/gstv4lsrc.h:
* sys/v4l/v4l_calls.c: (gst_v4l_open):
* sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_get_fps),
(gst_v4lsrc_get_fps_list), (gst_v4lsrc_buffer_new):
* sys/v4l/v4lsrc_calls.h:
Fractional framerates...
2005-11-22 23:46:31 +00:00
Jan Schmidt
f5ddd5037a Convert to fractional framerates.
Original commit message from CVS:
* ext/aalib/gstaasink.c: (gst_aasink_fixate):
* gst/debug/gstnavigationtest.c:
(gst_navigationtest_handle_src_event):
* gst/videofilter/gstvideofilter.c:
(gst_videofilter_format_get_structure), (gst_videofilter_setcaps),
(gst_videofilter_init):
* gst/videofilter/gstvideofilter.h:
Convert to fractional framerates.
2005-11-22 20:07:47 +00:00
Wim Taymans
af2acb6eea gst-libs/gst/audio/gstbaseaudiosink.c: And we provide a clock by default, of course...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init), (gst_base_audio_sink_set_clock),
(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
And we provide a clock by default, of course...
2005-11-22 18:54:56 +00:00
Wim Taymans
a3cb4d4937 gst-libs/gst/audio/gstaudioclock.c: This clock can be slaved to a master clock now.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init):
This clock can be slaved to a master clock now.

* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init), (gst_base_audio_sink_init),
(gst_base_audio_sink_dispose), (gst_base_audio_sink_provide_clock),
(gst_base_audio_sink_set_clock),
(gst_base_audio_sink_set_property),
(gst_base_audio_sink_get_property), (gst_base_audio_sink_preroll),
(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
* gst-libs/gst/audio/gstbaseaudiosink.h:
Handle slaving the internal clock to the clock selected in the
pipeline.
Add property to make the basesink not provide a clock.

* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_class_init), (gst_base_rtp_depayload_init),
(gst_base_rtp_depayload_wait):
* gst-libs/gst/rtp/gstbasertpdepayload.h:
We can use the clock in GstElement, no need to store it ourselves.
2005-11-22 18:32:09 +00:00
Jan Schmidt
0e82871285 Convert elements to use fractions for their framerate.
Original commit message from CVS:
* ext/libvisual/visual.c: (gst_visual_src_setcaps), (get_buffer),
(gst_visual_chain):
* ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
* ext/theora/theoradec.c: (theora_handle_type_packet):
* ext/theora/theoraenc.c: (theora_enc_sink_setcaps),
(theora_enc_chain):
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
* gst-libs/gst/video/video.c: (gst_video_frame_rate):
* gst-libs/gst/video/video.h:
* gst/ffmpegcolorspace/avcodec.h:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_caps_to_pixfmt):
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_set_caps):
* gst/videorate/gstvideorate.c: (gst_videorate_transformcaps),
(gst_videorate_setcaps), (gst_videorate_blank_data),
(gst_videorate_chain):
* gst/videotestsrc/gstvideotestsrc.c:
(gst_videotestsrc_src_fixate), (gst_videotestsrc_getcaps),
(gst_videotestsrc_parse_caps), (gst_videotestsrc_setcaps),
(gst_videotestsrc_event), (gst_videotestsrc_create):
* gst/videotestsrc/gstvideotestsrc.h:
* sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get),
(gst_ximagesink_setcaps), (gst_ximagesink_change_state),
(gst_ximagesink_get_times), (gst_ximagesink_init):
* sys/ximage/ximagesink.h:
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
(gst_xvimagesink_setcaps), (gst_xvimagesink_change_state),
(gst_xvimagesink_get_times), (gst_xvimagesink_init):
* sys/xvimage/xvimagesink.h:
Convert elements to use fractions for their framerate.
V4L elements to come later tonight.
2005-11-22 16:08:37 +00:00
Thomas Vander Stichele
08cd3b973f remove some deprecated functions
Original commit message from CVS:
remove some deprecated functions
2005-11-22 13:14:07 +00:00
Andy Wingo
20b31b4153 ext/ogg/gstoggdemux.c (gst_ogg_pad_submit_packet)
Original commit message from CVS:
2005-11-22  Andy Wingo  <wingo@pobox.com>

* ext/ogg/gstoggdemux.c (gst_ogg_pad_submit_packet)
(gst_ogg_demux_perform_seek):
* ext/theora/theoradec.c (theora_dec_sink_event):
* ext/vorbis/vorbisdec.c (vorbis_dec_sink_event): Run
update-funcnames.
2005-11-22 11:51:24 +00:00
Tim-Philipp Müller
c90f75526a Rename libgsttagedit to libgsttag (#322117).
Original commit message from CVS:
* ext/theora/Makefile.am:
* ext/vorbis/Makefile.am:
* gst-libs/gst/tag/Makefile.am:
* gst-plugins-base.spec.in:
Rename libgsttagedit to libgsttag (#322117).
2005-11-22 11:33:07 +00:00
Thomas Vander Stichele
1c3b6d42a9 gst-libs/gst/audio/audio.*: fix prototype - wondering why the test worked regardless
Original commit message from CVS:

* gst-libs/gst/audio/audio.c: (gst_audio_duration_from_pad_buffer):
* gst-libs/gst/audio/audio.h:
fix prototype - wondering why the test worked regardless
2005-11-21 23:51:45 +00:00
Thomas Vander Stichele
be5a7cd625 add a method that returns a proper GstClockTime
Original commit message from CVS:
add a method that returns a proper GstClockTime
2005-11-21 22:56:33 +00:00
Tim-Philipp Müller
aaf1e55bd2 gst-libs/gst/interfaces/xoverlay.*: Remove everything having to do with the desired size; add gst_x_overlay_prepare_x...
Original commit message from CVS:
* gst-libs/gst/interfaces/xoverlay.c: (gst_x_overlay_base_init),
(gst_x_overlay_got_xwindow_id), (gst_x_overlay_prepare_xwindow_id):
* gst-libs/gst/interfaces/xoverlay.h:
Remove everything having to do with the desired size; add
gst_x_overlay_prepare_xwindow_id() function; remove the
'have-xwindow-id' signal and make gst_x_overlay_got_xwindow_id()
post a message on the bus instead (#321816).
* sys/ximage/ximagesink.c: (gst_ximagesink_xoverlay_init):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps),
(gst_xvimagesink_xoverlay_init):
Remove desired size stuff (#321816).
2005-11-21 20:28:23 +00:00
Andy Wingo
3daa90cc7a gst-libs/gst/rtp/gstrtpbuffer.h: gst-libs/gst/rtp/gstrtpbuffer.c
Original commit message from CVS:
2005-11-21  Andy Wingo  <wingo@pobox.com>

patch from: Sebastien Cote <sebas642@yahoo.ca>

* gst-libs/gst/rtp/gstrtpbuffer.h:
* gst-libs/gst/rtp/gstrtpbuffer.c
(gst_rtpbuffer_get_payload_buffer): New function.
2005-11-21 19:12:51 +00:00
Wim Taymans
0f2336cff6 gst/: Segment update fix.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init), (gst_base_audio_sink_init),
(gst_base_audio_sink_provide_clock),
(gst_base_audio_sink_set_clock), (gst_base_audio_sink_render),
(gst_base_audio_sink_change_state):
* gst/audioresample/gstaudioresample.c:
Segment update fix.
2005-11-21 17:14:02 +00:00
Andy Wingo
f405e12b4a *.*: Ran scripts/update-macros. Oh yes.
Original commit message from CVS:
2005-11-21  Andy Wingo  <wingo@pobox.com>

* *.h:
* *.c: Ran scripts/update-macros. Oh yes.
2005-11-21 16:35:24 +00:00
Jan Schmidt
1cc82e9138 Rename gst_caps_structure_fixate_* to gst_structure_fixate_* (#322027)
Original commit message from CVS:
* ext/libvisual/visual.c: (get_buffer):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_fixate):
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_fixate_caps):
* gst/audioscale/gstaudioscale.c: (gst_audioscale_fixate):
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audiotestsrc_src_fixate):
* gst/sine/gstsinesrc.c: (gst_sinesrc_src_fixate):
* gst/videorate/gstvideorate.c: (gst_videorate_setcaps):
* gst/videoscale/gstvideoscale.c: (gst_videoscale_fixate_caps):
* gst/videotestsrc/gstvideotestsrc.c:
(gst_videotestsrc_src_fixate):
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_fixate):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_fixate):
Rename gst_caps_structure_fixate_* to gst_structure_fixate_*
(#322027)
2005-11-21 14:29:53 +00:00
Tim-Philipp Müller
213898e8dc Fixes for GST_FOURCC_FORMAT API change.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
(gst_riff_create_iavs_caps):
* gst-libs/gst/riff/riff-read.c: (gst_riff_parse_file_header),
(gst_riff_parse_strh), (gst_riff_parse_strf_vids),
(gst_riff_parse_info):
* gst/videotestsrc/videotestsrc.c: (paintinfo_find_by_structure):
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_set_caps):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps):
Fixes for GST_FOURCC_FORMAT API change.
2005-11-21 13:32:36 +00:00
Jan Schmidt
569286d058 gst-libs/gst/tag/: Remove obsolete vorbistag element and debug category.
Original commit message from CVS:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/tag/gstvorbistag.c:
(gst_tag_list_to_vorbiscomment_buffer):
Remove obsolete vorbistag element and debug category.

* gst/playback/gstplaybasebin.c: (check_queue):
Don't divide by 0 when queue-threshold is 0.

* sys/ximage/ximagesink.c: (gst_ximagesink_set_property):
Don't modify an existing pixel-aspect-ratio if we fail to read
a new one.
2005-11-20 17:11:06 +00:00
Wim Taymans
9edbf81fd2 gst-libs/gst/audio/gstbaseaudiosrc.c: Fix the audiosrc base class again, we did not unflush.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_change_state):
Fix the audiosrc base class again, we did not unflush.
2005-11-17 14:40:12 +00:00
Wim Taymans
99fb91493e gst-libs/gst/audio/gstbaseaudiosink.c: Set ringbuffer to non-flushing when going to PAUSED, set to flushing again whe...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_provide_clock),
(gst_base_audio_sink_change_state):
Set ringbuffer to non-flushing when going to PAUSED, set to
flushing again when going to READY.

* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_init),
(gst_ring_buffer_stop):
Start in flushing mode by default.
Don't set flushing in the _stop method, let the app call
this explicitly.
2005-11-16 16:48:35 +00:00
Julien Moutte
451645fd56 gst-libs/gst/video/: Add helper function needed for video sinks.
Original commit message from CVS:
2005-11-16  Julien MOUTTE  <julien@moutte.net>

* gst-libs/gst/video/gstvideosink.c:
(gst_video_sink_center_rect):
* gst-libs/gst/video/videosink.h: Add helper function needed
for video sinks.
2005-11-16 15:59:21 +00:00
Wim Taymans
8360581332 gst-libs/gst/audio/gstringbuffer.c: Set ringbuffer to flushing when stopping so that we don't block on wait_segment a...
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_stop):
Set ringbuffer to flushing when stopping so that we don't
block on wait_segment anymore and livelock.
2005-11-16 12:17:06 +00:00
Tim-Philipp Müller
9a16fb5d49 gst-libs/gst/rtp/gstbasertpdepayload.*: We need to send a newsegment event for each instance, not just for the first ...
Original commit message from CVS:
Reviewed by: Tim-Philipp Müller  <tim at centricular dot net>
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_init),
(gst_base_rtp_depayload_set_gst_timestamp):
* gst-libs/gst/rtp/gstbasertpdepayload.h:
We need to send a newsegment event for each instance, not
just for the first instance of this class (get rid of
static variable in function). (#321011).
2005-11-08 22:34:24 +00:00
Wim Taymans
b886b99345 gst-libs/gst/audio/gstbaseaudiosink.c: No need to do a typecheck.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
No need to do a typecheck.
2005-11-08 11:41:52 +00:00
Zeeshan Ali
61f87e1f1d Fixed a small problem.
Original commit message from CVS:
Fixed a small problem.
2005-11-04 18:03:07 +00:00
Zeeshan Ali
5194aac11a Don't sleep on the bench (system clock) when you have a nice comfortable bed (Gstreamer clock) to sleep on.
Original commit message from CVS:
Don't sleep on the bench (system clock) when you have a nice comfortable
bed (Gstreamer clock) to sleep on.
2005-11-03 22:49:38 +00:00
Zeeshan Ali
fd81fd44a6 Fixes some bugs in the depayloader's queuing/de-queueing code.
Original commit message from CVS:
Fixes some bugs in the depayloader's queuing/de-queueing code.
2005-11-02 10:23:38 +00:00
Zeeshan Ali
a93ba99167 Some random fixes, to fullfill the desires of thomas.
Original commit message from CVS:
Some random fixes, to fullfill the desires of thomas.
2005-10-31 15:32:54 +00:00
Zeeshan Ali
39daed4c1e Fixed the queueing algorithm.
Original commit message from CVS:
Fixed the queueing algorithm.
2005-10-31 13:29:06 +00:00
Zeeshan Ali
4771dfa4ca A small fix
Original commit message from CVS:
A small fix
2005-10-31 12:09:19 +00:00
Wim Taymans
d23d907a86 gst-libs/gst/audio/gstringbuffer.h: Don't break ABI.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.h:
Don't break ABI.

* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_caps_to_pixfmt):
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_set_caps):
Some more comments.
Handle missing required caps fields better.
2005-10-31 11:43:01 +00:00
Wim Taymans
09ca2ec93b gst-libs/gst/audio/: Add flushing mode to the ringbuffer so that it in all cases does not try to handle more audio. T...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_event), (gst_base_audio_sink_get_offset),
(gst_base_audio_sink_render):
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_open_device), (gst_ring_buffer_close_device),
(gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
(gst_ring_buffer_pause_unlocked), (gst_ring_buffer_pause),
(gst_ring_buffer_stop), (wait_segment), (gst_ring_buffer_commit),
(gst_ring_buffer_read):
* gst-libs/gst/audio/gstringbuffer.h:
Add flushing mode to the ringbuffer so that it in all cases does
not try to handle more audio. This makes sure it does not try to
block anymore when flushing and fixes a livelock.
2005-10-31 10:30:41 +00:00
Philippe Kalaf
f2ce71697b gst-libs/gst/rtp/gstbasedepayload.*: Minor cleanups
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasedepayload.c:
* gst-libs/gst/rtp/gstbasedepayload.h:
Minor cleanups
2005-10-27 22:14:02 +00:00
Zeeshan Ali
00e8471356 I'm too lazy to comment this
Original commit message from CVS:
*** empty log message ***
2005-10-26 21:42:21 +00:00
Zeeshan Ali
6c57193525 Fixes a small but nasty bug. The derived elements no longer segfaults on finalization.
Original commit message from CVS:
Fixes a small but nasty bug. The derived elements no longer segfaults on
finalization.
2005-10-26 20:00:46 +00:00
Zeeshan Ali
6b235f7ee3 Fixed a small mem-leak.
Original commit message from CVS:
Fixed a small mem-leak.
2005-10-26 14:19:21 +00:00
Zeeshan Ali
4417686358 Changed the C++ comments to C comments.
Original commit message from CVS:
Changed the C++ comments to C comments.
2005-10-26 13:52:42 +00:00
Zeeshan Ali
f6b0ca0572 The pad-template on the sinkpad should be set by the derived classes. Also added some usefull macros.
Original commit message from CVS:
The pad-template on the sinkpad should be set by the derived classes. Also added some usefull macros.
2005-10-25 17:20:55 +00:00
Wim Taymans
a878cbdfe1 gst-libs/gst/audio/gstbaseaudiosink.c: Remove g_print
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_get_offset), (gst_base_audio_sink_render):
Remove g_print
Use sync property from baseclass to disable sync.
2005-10-24 14:59:55 +00:00
Wim Taymans
cfadd55297 gst-libs/gst/audio/gstbaseaudiosink.c: Buffers with no timestamps get aligned with previous buffers or on underrun, p...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_get_offset), (gst_base_audio_sink_render):
Buffers with no timestamps get aligned with previous buffers or
on underrun, played ASAP.
2005-10-24 14:52:22 +00:00
Julien Moutte
69f68fa9f6 And here comes my change on caps for framerate and geometry range.
Original commit message from CVS:
2005-10-24  Julien MOUTTE  <julien@moutte.net>

* gst-libs/gst/video/video.h:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
* sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
And
here comes my change on caps for framerate and geometry range.
We are now accepting 1 to MAXINT for width and height, and from
0.0 to MAXDOUBLE for framerate. That allows duration less png
frames
to be blended correctly in videomixer.
2005-10-24 13:36:40 +00:00
Wim Taymans
7879080357 ext/: Fix old naming.
Original commit message from CVS:
* ext/theora/theoradec.c: (theora_dec_src_query),
(theora_dec_sink_event):
* ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
(vorbis_dec_sink_event), (vorbis_handle_identification_packet),
(vorbis_handle_data_packet):
* ext/vorbis/vorbisdec.h:
Fix old naming.

* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
Don't try to sync on buffers without a timestamp.
2005-10-21 15:14:36 +00:00
Wim Taymans
fc8ce00673 Bye bye buffer-frames.
Original commit message from CVS:
* check/elements/audioconvert.c:
* docs/libs/tmpl/gstaudio.sgml:
* docs/libs/tmpl/gstcolorbalance.sgml:
* ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
(vorbis_handle_identification_packet), (vorbis_handle_data_packet):
* ext/vorbis/vorbisenc.c: (raw_caps_factory):
* gst-libs/gst/audio/audio.c: (gst_audio_structure_set_int):
* gst-libs/gst/audio/audio.h:
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_parse_caps):
* gst/volume/gstvolume.c:
Bye bye buffer-frames.
2005-10-19 17:02:46 +00:00
Wim Taymans
efb6fcb802 ext/alsa/gstalsasink.c: Set handle to NULL.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_init),
(gst_alsasink_close):
Set handle to NULL.

* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_open_device), (gst_ring_buffer_close_device),
(gst_ring_buffer_acquire), (gst_ring_buffer_release),
(gst_ring_buffer_start), (gst_ring_buffer_pause),
(gst_ring_buffer_stop), (gst_ring_buffer_commit),
(gst_ring_buffer_read):
More debug info.
2005-10-18 11:07:26 +00:00
Tim-Philipp Müller
d022c250a1 gst-libs/gst/riff/riff-media.c: Add support for Indeo-3 (IV32).
Original commit message from CVS:
Reviewed by: Tim-Philipp Müller  <tim at centricular dot net>
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
(gst_riff_create_video_template_caps):
Add support for Indeo-3 (IV32).
2005-10-17 16:14:29 +00:00
Thomas Vander Stichele
fc0e077a93 gst/: doc updates
Original commit message from CVS:
* gst-libs/gst/video/video.c: (gst_video_get_size):
* gst/audiotestsrc/gstaudiotestsrc.c:
doc updates
2005-10-17 15:37:45 +00:00
Thomas Vander Stichele
6aff4166fa fix silly typo
Original commit message from CVS:
fix silly typo
2005-10-16 17:50:44 +00:00
Thomas Vander Stichele
4f8f42b0b6 restructure configure.ac, use correct libtool LDFLAGS, fix up defines
Original commit message from CVS:
restructure configure.ac, use correct libtool LDFLAGS, fix up defines
2005-10-16 13:54:44 +00:00
Tim-Philipp Müller
3b2a0751f9 Fix handling of GST_TAG_DATE, which is now of GST_TYPE_DATE.
Original commit message from CVS:
* ext/vorbis/vorbisenc.c:
* gst-libs/gst/tag/gstid3tag.c: (gst_tag_list_new_from_id3v1):
* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add),
(gst_tag_to_vorbis_comments):
Fix handling of GST_TAG_DATE, which is now of GST_TYPE_DATE.
2005-10-13 15:34:02 +00:00
Wim Taymans
1355459057 gst-libs/gst/audio/gstringbuffer.c: Don't assert on normal stuff.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_commit),
(gst_ring_buffer_read), (gst_ring_buffer_clear):
Don't assert on normal stuff.

* gst/playback/gstplaybin.c: (do_playbin_seek):
API fix.
2005-10-12 12:38:20 +00:00
Wim Taymans
5c17d94013 gst-libs/gst/audio/: Cleanups.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_commit),
(gst_ring_buffer_read):
Cleanups.
Commit and read from ringbuffer in samples rather than bytes.
2005-10-11 18:32:01 +00:00
Wim Taymans
0c71c6348f gst-libs/gst/audio/gstbaseaudiosink.c: Respect segment rate and accum when scheduling samples.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
Respect segment rate and accum when scheduling samples.
2005-10-11 17:31:48 +00:00
Wim Taymans
f13f1c0b3b check/generic/states.c: remove old property.
Original commit message from CVS:
* check/generic/states.c: (GST_START_TEST):
remove old property.

* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
(gst_ogg_demux_perform_seek):
* ext/theora/theoradec.c: (theora_dec_sink_event):
* ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
(vorbis_handle_data_packet):
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_set_gst_timestamp):
* gst/videorate/gstvideorate.c: (gst_videorate_event):
Update for newsegment API change.
2005-10-11 16:30:55 +00:00
Wim Taymans
81a09fc472 ext/alsa/gstalsasink.c: Also allow unsigned int.
Original commit message from CVS:
* ext/alsa/gstalsasink.c:
Also allow unsigned int.

* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create), (gst_base_audio_src_change_state):
Small cleanup
2005-10-10 17:04:24 +00:00
Philippe Kalaf
003cd58432 gst-libs/gst/rtp/rtpbasedepayload.c: Set timestamp and add queue delay to timestamp
Original commit message from CVS:
2005-10-09  Philippe Khalaf <burger@speedy.org>

* gst-libs/gst/rtp/rtpbasedepayload.c:
Set timestamp and add queue delay to timestamp
* gst-libs/gst/rtp/rtpbuffer.h:
Set correct payload type for h263
2005-10-09 21:32:34 +00:00
Wim Taymans
d920233a73 gst-libs/gst/audio/gstaudiosink.c: Only actually wait for the thread to be stopped if it's running.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_stop):
Only actually wait for the thread to be stopped if it's
running.
2005-10-08 12:02:08 +00:00
Wim Taymans
bd80afd2d1 gst-libs/gst/audio/gstbaseaudiosink.c: If we receive EOS we can start playback of what we had.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_event), (gst_base_audio_sink_render):
If we receive EOS we can start playback of what we had.
2005-10-08 11:47:52 +00:00
Edgard Lima
e846919fe9 gst-libs/gst/audio/: Fixed gstbaseaudiosrc adding ring buffer sync to it.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_event),
(gst_base_audio_src_create), (gst_base_audio_src_change_state):
* gst-libs/gst/audio/gstbaseaudiosrc.h:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_read):
patch from Edgard Lima <edgard.lima@indt.org.br>
Fixed gstbaseaudiosrc adding ring buffer sync to it.
2005-10-06 15:15:04 +00:00
Wim Taymans
a872aac9f8 ext/ogg/gstoggdemux.c: Report the FLOW_RETURN as string in the error message.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_loop):
Report the FLOW_RETURN as string in the error message.

* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_clear_all):
Don't assert when clearing an unnegotiated buffer.
2005-10-06 13:11:55 +00:00
Andy Wingo
721c97d438 gst-libs/gst/audio/gstringbuffer.c (gst_ring_buffer_clear)
Original commit message from CVS:
2005-10-02  Andy Wingo  <wingo@pobox.com>

* gst-libs/gst/audio/gstringbuffer.c (gst_ring_buffer_clear)
(gst_ring_buffer_prepare_read):
* gst-libs/gst/audio/gstaudiosink.c (audioringbuffer_thread_func):
Demote to LOG.
2005-10-02 15:58:57 +00:00
Wim Taymans
79be8760f0 gst-libs/gst/audio/: get_clock -> provide_clock
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init),
(gst_base_audio_sink_provide_clock), (gst_base_audio_sink_render):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init),
(gst_base_audio_src_provide_clock):
get_clock -> provide_clock
2005-09-28 13:41:29 +00:00
Wim Taymans
b17856db22 Fix sync again. Moved sample alignment to basesink.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_reset):
* gst-libs/gst/audio/gstaudiosink.c:
(gst_audioringbuffer_class_init), (audioringbuffer_thread_func),
(gst_audioringbuffer_stop):
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_event), (gst_base_audio_sink_render),
(gst_base_audio_sink_change_state):
* gst-libs/gst/audio/gstbaseaudiosink.h:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_stop),
(gst_ring_buffer_set_sample), (gst_ring_buffer_clear_all),
(gst_ring_buffer_commit), (gst_ring_buffer_read):
* gst-libs/gst/audio/gstringbuffer.h:
Fix sync again. Moved sample alignment to basesink.
2005-09-24 13:06:03 +00:00
Thomas Vander Stichele
272aad79bb add/fix docs
Original commit message from CVS:

* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* gst/volume/gstvolume.c:
add/fix docs
* gst-libs/gst/audio/audio.c: (gst_audio_frame_byte_size):
* gst-libs/gst/audio/audio.h:
add conversion macros for frames <-> clocktime
2005-09-23 18:14:54 +00:00
Wim Taymans
9798126a0b gst-libs/gst/rtp/gstbasertppayload.*: Added max-ptime to control amount of data in the rtp packets.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_class_init), (gst_basertppayload_init),
(gst_basertppayload_setcaps), (gst_basertppayload_chain),
(gst_basertppayload_set_options), (gst_basertppayload_set_outcaps),
(gst_basertppayload_is_filled), (gst_basertppayload_push),
(gst_basertppayload_set_property),
(gst_basertppayload_get_property),
(gst_basertppayload_change_state):
* gst-libs/gst/rtp/gstbasertppayload.h:
Added max-ptime to control amount of data in the rtp packets.
2005-09-22 14:13:04 +00:00
Wim Taymans
9bbe8af0c3 gst-libs/gst/rtp/gstbasertppayload.c: Allow 0 ssrc too.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_class_init), (gst_basertppayload_init),
(gst_basertppayload_setcaps), (gst_basertppayload_chain),
(gst_basertppayload_set_options), (gst_basertppayload_set_outcaps),
(gst_basertppayload_push), (gst_basertppayload_get_property),
(gst_basertppayload_change_state):
Allow 0 ssrc too.
2005-09-21 11:49:37 +00:00
Wim Taymans
597ace35b5 gst-libs/gst/rtp/gstbasertppayload.*: Added property to configure sequence number offsets.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_class_init), (gst_basertppayload_init),
(gst_basertppayload_setcaps), (gst_basertppayload_chain),
(gst_basertppayload_set_options), (gst_basertppayload_set_outcaps),
(gst_basertppayload_push), (gst_basertppayload_set_property),
(gst_basertppayload_get_property),
(gst_basertppayload_change_state):
* gst-libs/gst/rtp/gstbasertppayload.h:
Added property to configure sequence number offsets.
2005-09-20 13:34:02 +00:00
Wim Taymans
25e6dc6013 gst-libs/gst/rtp/gstbasertppayload.*: Make timestamp offset configurable.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_class_init), (gst_basertppayload_init),
(gst_basertppayload_setcaps), (gst_basertppayload_chain),
(gst_basertppayload_set_options), (gst_basertppayload_set_outcaps),
(gst_basertppayload_push), (gst_basertppayload_set_property),
(gst_basertppayload_get_property),
(gst_basertppayload_change_state):
* gst-libs/gst/rtp/gstbasertppayload.h:
Make timestamp offset configurable.
2005-09-20 11:50:20 +00:00
Tim-Philipp Müller
f09d105f14 gst-libs/gst/interfaces/propertyprobe.c: Fix wrong macro usage; it's G_OBJECT_GET_CLASS(obj) or ot G_OBJECT_CLASS(obj...
Original commit message from CVS:
Reviewed by: Tim-Philipp Müller  <tim at centricular dot net>
* gst-libs/gst/interfaces/propertyprobe.c:
(gst_property_probe_probe_property_name),
(gst_property_probe_needs_probe_name),
(gst_property_probe_get_values_name),
(gst_property_probe_probe_and_get_values_name):
Fix wrong macro usage; it's G_OBJECT_GET_CLASS(obj) or
G_OBJECT_CLASS(klass), not G_OBJECT_CLASS(obj). (#316571)
2005-09-19 17:25:00 +00:00
Wim Taymans
e1c64e3b2f gst-libs/gst/rtp/gstbasertppayload.c: Posting ERROR and WARNING messages is good.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_class_init), (gst_basertppayload_init),
(gst_basertppayload_setcaps), (gst_basertppayload_chain),
(gst_basertppayload_set_options), (gst_basertppayload_set_outcaps),
(gst_basertppayload_push), (gst_basertppayload_get_property),
(gst_basertppayload_change_state):
Posting ERROR and WARNING messages is good.
2005-09-19 14:23:33 +00:00
Wim Taymans
16c30a244b gst-libs/gst/rtp/gstbasertpdepayload.c: This one was not supposed to go in.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_add_to_queue),
(gst_base_rtp_depayload_push),
(gst_base_rtp_depayload_set_gst_timestamp),
(gst_base_rtp_depayload_queue_release):
This one was not supposed to go in.
2005-09-19 11:31:29 +00:00
Wim Taymans
929b8afd16 check/pipelines/simple_launch_lines.c: Fix for bus API.
Original commit message from CVS:
* check/pipelines/simple_launch_lines.c: (run_pipeline):
Fix for bus API.

* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_chain),
(gst_base_rtp_depayload_add_to_queue),
(gst_base_rtp_depayload_push),
(gst_base_rtp_depayload_set_gst_timestamp),
(gst_base_rtp_depayload_queue_release):
Some cleanups.

* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_class_init), (gst_basertppayload_init),
(gst_basertppayload_setcaps), (gst_basertppayload_set_options),
(gst_basertppayload_set_outcaps), (gst_basertppayload_push),
(gst_basertppayload_get_property),
(gst_basertppayload_change_state):
Added debugging category.
2005-09-19 11:24:46 +00:00
Wim Taymans
022f6698f0 gst-libs/gst/rtp/: Added rtp payloader base class.
Original commit message from CVS:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_get_type), (gst_basertppayload_base_init),
(gst_basertppayload_class_init), (gst_basertppayload_init),
(gst_basertppayload_finalize), (gst_basertppayload_setcaps),
(gst_basertppayload_chain), (gst_basertppayload_set_options),
(gst_basertppayload_set_outcaps), (gst_basertppayload_push),
(gst_basertppayload_set_property),
(gst_basertppayload_get_property),
(gst_basertppayload_change_state):
* gst-libs/gst/rtp/gstbasertppayload.h:
Added rtp payloader base class.
2005-09-15 13:50:05 +00:00
David Schleef
cb8927cb92 Fixes for changes in registry API.
Original commit message from CVS:
* check/generic/states.c:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind):
* gst/playback/gstdecodebin.c: (gst_decode_bin_init):
Fixes for changes in registry API.
* configure.ac: Only export gst_plugins_desc.  Add -no-undefined
to GST_PLUGIN_LDFLAGS.
* ext/libvisual/visual.c: Make the library shut up.
* gst-libs/gst/audio/audio.c: Don't define a plugin in a library.
* gst-libs/gst/audio/gstaudiofilter.c: same
2005-09-15 06:59:36 +00:00
Stefan Kost
1aa698fefa gst/: fixing lost sync, some more debugging
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_commit):
* gst/volume/gstvolume.c: (gst_volume_class_init),
(volume_transform):
fixing lost sync, some more debugging
2005-09-08 22:42:11 +00:00
Stefan Kost
896950ce84 gsttaginterface.h -> gsttagsetter.h
Original commit message from CVS:
* ext/vorbis/vorbisenc.c:
* gst-libs/gst/tag/gstvorbistag.c:
gsttaginterface.h -> gsttagsetter.h
2005-09-07 13:55:08 +00:00
Andy Wingo
6665c3084c All plugins updated for element state changes.
Original commit message from CVS:
2005-09-02  Andy Wingo  <wingo@pobox.com>

* All plugins updated for element state changes.
2005-09-02 15:43:18 +00:00
Wim Taymans
44cc3421a0 gst-libs/gst/audio/gstbaseaudiosink.c: Resync if the buffer timestamps drift more than a 10th of a second.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
Resync if the buffer timestamps drift more than a 10th
of a second.
2005-08-31 10:57:35 +00:00
Andy Wingo
c32721723b Updates for two-arg init from GST_BOILERPLATE_FULL.
Original commit message from CVS:
2005-08-28  Andy Wingo  <wingo@pobox.com>

* Updates for two-arg init from GST_BOILERPLATE_FULL.
2005-08-28 17:52:45 +00:00
Wim Taymans
7824216cef ext/ogg/gstoggdemux.c: Parse seeking events better.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query),
(gst_ogg_pad_event), (gst_ogg_demux_factory_filter),
(gst_ogg_pad_submit_packet), (gst_ogg_chain_new),
(gst_ogg_demux_init), (gst_ogg_demux_perform_seek),
(gst_ogg_demux_collect_chain_info), (gst_ogg_demux_collect_info),
(gst_ogg_demux_chain), (gst_ogg_demux_loop), (gst_ogg_print):
Parse seeking events better.
Unref static caps.
Generate correct newsegment events, fixes seeking in live oggs.

* ext/theora/theoradec.c: (theora_dec_src_query),
(theora_dec_src_event), (theora_dec_src_getcaps),
(theora_dec_sink_event), (theora_dec_push), (theora_dec_chain):
Use newsegment values to report correct play time.

* ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
(vorbis_dec_src_event), (vorbis_dec_sink_event):
* ext/vorbis/vorbisdec.h:
Parse and use newsegment values to report correct play time.

* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_event), (gst_base_audio_sink_render):
Clear ringbuffer on flush.
Use newsegment values to calculate playback time.

* sys/ximage/ximagesink.c: (gst_ximagesink_get_times):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_times):
Basesink does newsegment calculations for us now.
2005-08-24 18:04:45 +00:00
Wim Taymans
5ac2327f05 gst-libs/gst/audio/gstringbuffer.*: Added function to clear the ringbuffer.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_set_sample),
(gst_ring_buffer_clear_all):
* gst-libs/gst/audio/gstringbuffer.h:
Added function to clear the ringbuffer.
2005-08-24 11:29:10 +00:00
Andy Wingo
7afb104567 gst-libs/gst/audio/gstbaseaudiosrc.c
Original commit message from CVS:
2005-08-23  Andy Wingo  <wingo@pobox.com>

* gst-libs/gst/audio/gstbaseaudiosrc.c
(gst_base_audio_src_fixate): Only fixate endianness if it is
present in the caps.
2005-08-23 13:29:17 +00:00
Andy Wingo
13b122a106 gst-libs/gst/audio/gstaudiosrc.*: Implement open_device and close_device in the ring buffer, like gstaudiosink.
Original commit message from CVS:
2005-08-22  Andy Wingo  <wingo@pobox.com>

* gst-libs/gst/audio/gstaudiosrc.h:
* gst-libs/gst/audio/gstaudiosrc.c: Implement open_device and
close_device in the ring buffer, like gstaudiosink.

* ext/alsa/gstalsamixer.h:
* ext/alsa/gstalsamixer.c: Not a GObject any more. Include a nifty
macro to implement the interface without much code. Cleanups.

* ext/alsa/gstalsasrc.h:
* ext/alsa/gstalsasrc.c: Be a mixer. Open device and mixer in
READY.

* ext/alsa/Makefile.am: Add new files.
* ext/alsa/gstalsamixerelement.c:
* ext/alsa/gstalsamixerelement.c: Split element code out from
mixer code so that alsasrc can be a mixer too.
2005-08-22 15:11:31 +00:00
Wim Taymans
7667a989d3 gst-libs/gst/rtp/gstbasertpdepayload.*: Fix for RTPBuffer changes.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_chain),
(gst_base_rtp_depayload_add_to_queue),
(gst_base_rtp_depayload_push),
(gst_base_rtp_depayload_queue_release):
* gst-libs/gst/rtp/gstbasertpdepayload.h:
Fix for RTPBuffer changes.

* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtpbuffer_allocate_data),
(gst_rtpbuffer_new_take_data), (gst_rtpbuffer_new_copy_data),
(gst_rtpbuffer_new_allocate), (gst_rtpbuffer_new_allocate_len),
(gst_rtpbuffer_calc_header_len), (gst_rtpbuffer_calc_packet_len),
(gst_rtpbuffer_calc_payload_len), (gst_rtpbuffer_validate_data),
(gst_rtpbuffer_validate), (gst_rtpbuffer_set_packet_len),
(gst_rtpbuffer_get_packet_len), (gst_rtpbuffer_get_version),
(gst_rtpbuffer_set_version), (gst_rtpbuffer_get_padding),
(gst_rtpbuffer_set_padding), (gst_rtpbuffer_pad_to),
(gst_rtpbuffer_get_extension), (gst_rtpbuffer_set_extension),
(gst_rtpbuffer_get_ssrc), (gst_rtpbuffer_set_ssrc),
(gst_rtpbuffer_get_csrc_count), (gst_rtpbuffer_get_csrc),
(gst_rtpbuffer_set_csrc), (gst_rtpbuffer_get_marker),
(gst_rtpbuffer_set_marker), (gst_rtpbuffer_get_payload_type),
(gst_rtpbuffer_set_payload_type), (gst_rtpbuffer_get_seq),
(gst_rtpbuffer_set_seq), (gst_rtpbuffer_get_timestamp),
(gst_rtpbuffer_set_timestamp), (gst_rtpbuffer_get_payload_len),
(gst_rtpbuffer_get_payload):
* gst-libs/gst/rtp/gstrtpbuffer.h:
Don't subclass GstBuffer but add methods and helper functions
to construct and manipulate RTP packets in regular GstBuffers.
2005-08-18 10:23:54 +00:00
Wim Taymans
4e3b19e5fb gst-libs/gst/audio/gstbaseaudiosrc.c: Open and close device in READY<->NULL state change.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_change_state):
Open and close device in READY<->NULL state change.
2005-08-16 15:53:59 +00:00
Philippe Kalaf
96a0b1b9b9 gst-libs/gst/rtp/gstbasertpdepayload.*: Made a thread to release the queue.
Original commit message from CVS:
2005-08-12  Philippe Khalaf <burger@speedy.org>
* gst-libs/gst/rtp/gstbasertpdepayload.c:
* gst-libs/gst/rtp/gstbasertpdepayload.h:
Made a thread to release the queue.
Removed timestamp conversion for now.
2005-08-12 13:34:56 +00:00
Philippe Kalaf
b50d3fe5f6 gst-libs/gst/rtp/gstbasertpdepayload.*: Added rtp timestamp -> gst timestamp conversion.
Original commit message from CVS:
2005-08-10  Philippe Khalaf <burger@speedy.org>
* gst-libs/gst/rtp/gstbasertpdepayload.c:
* gst-libs/gst/rtp/gstbasertpdepayload.h:
Added rtp timestamp -> gst timestamp conversion.
Fixed several problems with queue.
2005-08-10 20:52:37 +00:00
Tim-Philipp Müller
b9b56ce7d3 gst-libs/gst/: Add padding (you will need to rebuild gst-plugins-base, gst-plugins and all applications afterwards!)
Original commit message from CVS:
* gst-libs/gst/audio/gstaudioclock.h:
* gst-libs/gst/audio/gstaudiofilter.h:
* gst-libs/gst/audio/gstaudiosink.h:
* gst-libs/gst/audio/gstaudiosrc.h:
* gst-libs/gst/audio/gstbaseaudiosink.h:
* gst-libs/gst/audio/gstbaseaudiosrc.h:
* gst-libs/gst/audio/gstringbuffer.h:
* gst-libs/gst/net/gstnetbuffer.h:
* gst-libs/gst/rtp/gstbasertpdepayload.h:
* gst-libs/gst/rtp/gstrtpbuffer.h:
Add padding (you will need to rebuild gst-plugins-base,
gst-plugins and all applications afterwards!)
2005-08-09 17:29:40 +00:00
Tim-Philipp Müller
822e77203a gst-libs/gst/riff/riff-read.c: Fix bug in debug message and add some more debug messages.
Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk),
(gst_riff_parse_chunk):
Fix bug in debug message and add some more debug messages.
2005-08-09 16:59:21 +00:00
Edward Hervey
b060089ac9 gst-libs/gst/riff/riff-media.c: backported updates since branch
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
backported updates since branch
2005-08-08 16:58:29 +00:00
Andy Wingo
69d36f02ce gst-libs/gst/audio/gstbaseaudiosink.c
Original commit message from CVS:
2005-08-08  Andy Wingo  <wingo@pobox.com>

* gst-libs/gst/audio/gstbaseaudiosink.c
(gst_base_audio_sink_change_state): Open the device in NULL->READY
like good elements should. Close on READY->NULL too.

* gst-libs/gst/audio/gstaudiosink.c
(gst_audioringbuffer_open_device,
(gst_audioringbuffer_close_device, gst_audioringbuffer_acquire)
(gst_audioringbuffer_release): Updates for new ring buffer API,
hook into the new audio sink api.

* gst-libs/gst/audio/gstaudiosink.h (GstAudioSinkClass.open)
(GstAudioSinkClass.close): Just open and close the device -- no
resource allocation or configuration.
(GstAudioSinkClass.prepare, GstAudioSinkClass.unprepare): New
vmethods, handle device setup and resource allocation.

* ext/alsa/gstalsasink.c (gst_alsasink_open, gst_alsasink_close)
(gst_alsasink_prepare, gst_alsasink_unprepare): Update for new
base class API.

* gst-libs/gst/audio/gstringbuffer.h
(GstRingBufferClass.open_device, GstRingBufferClass.close_device):
New vmethods.

* gst-libs/gst/audio/gstringbuffer.c (gst_ring_buffer_open_device)
(gst_ring_buffer_close_device, gst_ring_buffer_device_is_open):
New API functions. The device should be opened before acquiring
and closed after releasing.
2005-08-08 16:42:10 +00:00
Tim-Philipp Müller
074b6a3b64 gst-libs/gst/interfaces/mixer.h: Reset padding to GST_PADDING.
Original commit message from CVS:
* gst-libs/gst/interfaces/mixer.h:
Reset padding to GST_PADDING.
2005-08-08 14:13:59 +00:00
Ronald S. Bultje
6550429258 gst-libs/gst/gconf/gconf.*: Fix some Andy Problem [tm].
Original commit message from CVS:
* gst-libs/gst/gconf/gconf.c:
* gst-libs/gst/gconf/gconf.h:
Fix some Andy Problem [tm].
2005-08-05 15:33:19 +00:00
Wim Taymans
78b9a84efa gst-libs/gst/audio/gstbaseaudiosrc.c: More compilation fixen.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_event):
More compilation fixen.
2005-07-27 19:13:27 +00:00
Wim Taymans
50b9b8acc4 gst-libs/gst/audio/gstbaseaudiosink.c: Fix compilation.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_event), (gst_base_audio_sink_render),
(gst_base_audio_sink_create_ringbuffer),
(gst_base_audio_sink_change_state):
Fix compilation.
2005-07-27 19:10:20 +00:00
Thomas Vander Stichele
9449780dc3 forward port from 0.9 and enable videoflip now that it works
Original commit message from CVS:
forward port from 0.9 and enable videoflip now that it works
2005-07-25 14:06:15 +00:00
Wim Taymans
e2da9961d9 ext/ogg/gstoggdemux.c: Generate correct disconts for live chained oggs.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_event),
(gst_ogg_pad_internal_chain), (gst_ogg_pad_typefind),
(gst_ogg_demux_chain_elem_pad), (gst_ogg_demux_queue_data),
(gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
(gst_ogg_pad_submit_page), (gst_ogg_chain_new),
(gst_ogg_demux_init), (gst_ogg_demux_activate_chain),
(gst_ogg_demux_perform_seek), (gst_ogg_demux_collect_chain_info),
(gst_ogg_demux_collect_info), (gst_ogg_demux_chain),
(gst_ogg_demux_send_event), (gst_ogg_demux_loop):
Generate correct disconts for live chained oggs.

* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render),
(gst_base_audio_sink_create_ringbuffer),
(gst_base_audio_sink_change_state):
Handle discont math correctly.

* gst/playback/gstplaybin.c: (add_sink):
Some small debug cleanup.
2005-07-21 17:25:40 +00:00
Ronald S. Bultje
7795794bf0 Fixes for API changes in core.
Original commit message from CVS:
* ext/ogg/gstoggmux.c: (gst_ogg_mux_get_headers),
(gst_ogg_mux_set_header_on_caps):
* ext/theora/theoraenc.c: (theora_set_header_on_caps):
* ext/vorbis/vorbisenc.c: (gst_vorbisenc_set_header_on_caps):
* ext/vorbis/vorbisparse.c: (vorbis_parse_set_header_on_caps):
* gst-libs/gst/audio/multichannel.c:
(gst_audio_set_channel_positions),
(gst_audio_set_structure_channel_positions_list):
* gst/playback/gstdecodebin.c: (dynamic_create):
* gst/playback/gstplaybasebin.c: (setup_source), (mute_group_type):
* gst/playback/gststreaminfo.c: (gst_stream_info_set_mute):
Fixes for API changes in core.
2005-07-20 17:16:54 +00:00
Ronald S. Bultje
950ecc5d5f Remove gconf stuff, use gconf elements instead from now on.
Original commit message from CVS:
* configure.ac:
* gst-libs/gst/Makefile.am:
* gst-libs/gst/gconf/.cvsignore:
* gst-libs/gst/gconf/Makefile.am:
* gst-libs/gst/gconf/test-gconf.c:
* pkgconfig/Makefile.am:
* pkgconfig/gstreamer-gconf-uninstalled.pc.in:
* pkgconfig/gstreamer-gconf.pc.in:
Remove gconf stuff, use gconf elements instead from now on.
2005-07-20 10:12:34 +00:00
Wim Taymans
ee345636bc gst-libs/gst/audio/: Make sure the audio clock always returns an increasing value.
Original commit message from CVS:
* gst-libs/gst/audio/TODO:
* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
(gst_audio_clock_get_internal_time):
* gst-libs/gst/audio/gstaudioclock.h:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_init), (gst_base_audio_sink_dispose),
(gst_base_audio_sink_get_time), (gst_base_audio_sink_event),
(gst_base_audio_sink_render),
(gst_base_audio_sink_create_ringbuffer),
(gst_base_audio_sink_change_state):
Make sure the audio clock always returns an increasing value.
2005-07-20 09:08:05 +00:00
Wim Taymans
c84a6b964f gst-libs/gst/audio/gstbaseaudiosink.c: Align samples even if we have roundoff errors in the timestamp conversion.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
Align samples even if we have roundoff errors in the
timestamp conversion.
2005-07-16 17:13:35 +00:00
Wim Taymans
82dc411e33 Updated seek example.
Original commit message from CVS:
* docs/libs/tmpl/gstringbuffer.sgml:
* examples/seeking/seek.c: (make_vorbis_theora_pipeline),
(query_rates), (query_positions_elems), (query_positions_pads),
(update_scale), (do_seek):
Updated seek example.

* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
(gst_ogg_pad_submit_page), (gst_ogg_demux_activate_chain),
(gst_ogg_demux_find_chains), (gst_ogg_demux_send_event),
(gst_ogg_demux_loop):
Push out correct discont values.

* ext/theora/theoradec.c: (theora_dec_src_convert),
(theora_dec_sink_convert), (theora_dec_src_getcaps),
(theora_dec_sink_event), (theora_handle_type_packet),
(theora_handle_header_packet), (theora_dec_push),
(theora_handle_data_packet), (theora_dec_chain),
(theora_dec_change_state):
Better timestamping.

* ext/vorbis/vorbisdec.c: (gst_vorbis_dec_init),
(vorbis_dec_sink_event), (vorbis_dec_push),
(vorbis_handle_data_packet), (vorbis_dec_chain):
* ext/vorbis/vorbisdec.h:
Better timestamping.

* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_get_time), (gst_base_audio_sink_get_times),
(gst_base_audio_sink_event), (gst_base_audio_sink_render):
Handle syncing on timestamps instead of sample offsets. Make
use of DISCONT values as described in design docs.

* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_get_time):
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_acquire),
(gst_ring_buffer_set_sample), (gst_ring_buffer_commit),
(gst_ring_buffer_read):
* gst-libs/gst/audio/gstringbuffer.h:
* sys/ximage/ximagesink.c: (gst_ximagesink_get_times),
(gst_ximagesink_show_frame):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_times):
Correcly convert buffer timestamp to stream time.
2005-07-16 14:47:27 +00:00
Philippe Kalaf
dc0610faff gst-libs/gst/rtp gst-libs/gst/rtp/gstbasertpdepayload.c gst-libs/gst/rtp/gstbasertpdepayload.h gst-libs/gst/rtp/gstrt...
Original commit message from CVS:
* gst-libs/gst/rtp
* gst-libs/gst/rtp/gstbasertpdepayload.c
* gst-libs/gst/rtp/gstbasertpdepayload.h
* gst-libs/gst/rtp/gstrtpbuffer.c
* gst-libs/gst/rtp/gstrtpbuffer.h
* gst-libs/gst/rtp/Makefile.am
* gst-libs/gst/rtp/README

Support libs for RTP. Basicaly this add a GstRTPBuffer (extended GstBuffer) and
a Depayloader Base class that shall be used by payload specific depayloaders.
2005-07-14 10:29:30 +00:00
Thomas Vander Stichele
d143d256a6 more autistic cleanliness in functions/names/defines
Original commit message from CVS:
more autistic cleanliness in functions/names/defines
2005-07-14 09:35:11 +00:00
Thomas Vander Stichele
5852e82a04 install but don't dist the enumtypes
Original commit message from CVS:
install but don't dist the enumtypes
2005-07-13 19:28:39 +00:00
Thomas Vander Stichele
3489636a8d install the enumtypes header because audio plugins in other modules need it
Original commit message from CVS:
install the enumtypes header because audio plugins in other modules need it
2005-07-13 19:25:23 +00:00
Thomas Vander Stichele
1ea0574af4 make GST_PLUGIN_LDFLAGS only be flags; GST_LIBS should be added manually to each Makefile.am so we are sure it goes
Original commit message from CVS:
make GST_PLUGIN_LDFLAGS only be flags; GST_LIBS should be
added manually to each Makefile.am so we are sure it goes
*last* and doesn't add -L flags before linking in libs of our
own, like, say, internal .la libs, that then accidentally pick
up the installed copy.
2005-07-13 17:58:07 +00:00
Thomas Vander Stichele
9042241fd1 remove whitespace
Original commit message from CVS:
remove whitespace
2005-07-12 23:15:35 +00:00
Thomas Vander Stichele
9e8a11d3ce use overridable ERROR_CFLAGS; more macro splitting
Original commit message from CVS:
use overridable ERROR_CFLAGS; more macro splitting
2005-07-10 12:03:58 +00:00
Wim Taymans
ceb88a7777 Added audiosource base classes.
Original commit message from CVS:
Added audiosource base classes.
Ported alsasrc, still very basic.
2005-07-06 15:27:17 +00:00
Andy Wingo
e4180644b1 Many files: Null if we got it....
Original commit message from CVS:
2005-07-05  Andy Wingo  <wingo@pobox.com>

* Many files: Null if we got it....
2005-07-05 11:08:56 +00:00
Andy Wingo
68eeef9614 Way, way, way too many files: Remove crack comment from the 2000 era.
Original commit message from CVS:
2005-07-05  Andy Wingo  <wingo@pobox.com>

* Way, way, way too many files:
Remove crack comment from the 2000 era.
2005-07-05 10:51:45 +00:00
Thomas Vander Stichele
ab4fbc5655 adding mixer docs
Original commit message from CVS:
adding mixer docs
2005-06-30 15:34:38 +00:00
Thomas Vander Stichele
c600abef63 commit
Original commit message from CVS:
commit
2005-06-30 14:51:33 +00:00
Thomas Vander Stichele
c5d3e65e86 adding interfaces.h
Original commit message from CVS:
adding interfaces.h
2005-06-30 12:09:26 +00:00
Thomas Vander Stichele
bb8da77fa8 ignore more
Original commit message from CVS:
ignore more
2005-06-30 12:01:41 +00:00
Thomas Vander Stichele
eeb7984bfd remove old ones
Original commit message from CVS:
remove old ones
2005-06-30 12:00:42 +00:00
Thomas Vander Stichele
5014a9eee6 fold all interfaces in one interfaces dir, preserving CVS history
Original commit message from CVS:
fold all interfaces in one interfaces dir, preserving CVS history
2005-06-30 11:58:40 +00:00
Ronald S. Bultje
a8bff2dd5e gst-libs/gst/riff/: Add gst_riff_init() to initialize the debug category, instead of plugin_init(). Port riff-media.[...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
(gst_riff_create_audio_caps), (gst_riff_create_iavs_caps),
(gst_riff_create_video_template_caps),
(gst_riff_create_audio_template_caps),
(gst_riff_create_iavs_template_caps):
* gst-libs/gst/riff/riff-media.h:
* gst-libs/gst/riff/riff-read.h:
* gst-libs/gst/riff/riff.c: (gst_riff_init):
Add gst_riff_init() to initialize the debug category, instead
of plugin_init(). Port riff-media.[ch] from -THREADED to HEAD.
2005-06-30 08:59:30 +00:00
Wim Taymans
2e2623748d gst-libs/gst/audio/: Fix compilation error.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_init):
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_baseaudiosink_class_init), (gst_baseaudiosink_dispose),
(gst_baseaudiosink_change_state):
* gst-libs/gst/audio/gstbaseaudiosink.h:
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ringbuffer_set_callback):
Fix compilation error.
Ringbuffer starts out as not running.
Free our clock in dispose.
When releasing the ringbuffer we need to renegotiate so
clear the pad caps.
2005-06-29 11:17:33 +00:00
Thomas Vander Stichele
36d0c9ce29 reinstate plugin docs
Original commit message from CVS:
reinstate plugin docs
2005-06-29 10:56:25 +00:00
Andy Wingo
d0bc038021 *.c: Don't cast to GstObject before reffing/unreffing.
Original commit message from CVS:
2005-06-28  Andy Wingo  <wingo@pobox.com>

* *.c: Don't cast to GstObject before reffing/unreffing.
2005-06-28 10:16:13 +00:00
Jan Schmidt
2255ffd384 gst-libs/gst/audio/gstaudiosink.c: Set the worker thread's running flag to TRUE before starting the thread.
Original commit message from CVS:
2005-06-25  Jan Schmidt  <thaytan@mad.scientist.com>
* gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_acquire):
Set the worker thread's running flag to TRUE before starting the
thread.
* gst/playback/gstdecodebin.c: (gst_decode_bin_init):
Catch a failure to add typefind to the bin.
2005-06-24 16:15:25 +00:00
Andy Wingo
9079621594 gst-libs/gst/net/Makefile.am (lib_LTLIBRARIES): Install gstnet.
Original commit message from CVS:
2005-06-09  Andy Wingo  <wingo@pobox.com>

* gst-libs/gst/net/Makefile.am (lib_LTLIBRARIES): Install gstnet.
2005-06-09 15:01:54 +00:00
Andy Wingo
cd8caf2f9f Add gstnet to build.
Original commit message from CVS:
2005-06-09  Andy Wingo  <wingo@pobox.com>

* configure.ac:
* gst-libs/gst/Makefile.am:
* gst-libs/gst/net/Makefile.am:
Add gstnet to build.
2005-06-09 08:54:45 +00:00
Andy Wingo
55d437af1c gst/: Ghost pad API fixes.
Original commit message from CVS:
2005-06-09  Andy Wingo  <wingo@pobox.com>

* gst-libs/gst/gconf/gconf.c:
* gst/playback/test.c:
* gst/playback/gstplaybin.c (gen_video_element): Ghost pad API
fixes.

* gst/audioconvert/gstaudioconvert.c: RPAD fixes.

* ext/theora/theoraenc.c (theora_enc_chain):
* ext/theora/theoradec.c (theora_handle_data_packet): GCC4 fixes.

* ext/ogg/gstoggdemux.c (GstOggPad): Derive from GstPad, not
RealPad.
2005-06-08 22:18:05 +00:00
Wim Taymans
ec4f41ed25 Added net stuff, version net lib.
Original commit message from CVS:
* gst-libs/gst/net/Makefile.am:
* pkgconfig/gstreamer-libs-uninstalled.pc.in:
* pkgconfig/gstreamer-libs.pc.in:
Added net stuff, version net lib.
2005-06-02 14:01:22 +00:00
Wim Taymans
d014bb6f43 gst/: Bufferalloc changes.
Original commit message from CVS:
* gst/effectv/gstquark.c: (gst_quarktv_chain):
* gst/goom/gstgoom.c: (gst_goom_chain):
* gst/videobox/Makefile.am:
* gst/videobox/gstvideobox.c: (gst_video_box_class_init),
(gst_video_box_init), (gst_video_box_sink_setcaps),
(gst_video_box_chain):
* gst/videofilter/gstvideofilter.c: (gst_videofilter_chain):
* gst/videorate/gstvideorate.c: (gst_videorate_class_init),
(gst_videorate_getcaps), (gst_videorate_setcaps),
(gst_videorate_init), (gst_videorate_event), (gst_videorate_chain),
(gst_videorate_change_state):
Bufferalloc changes.
2005-06-02 10:03:23 +00:00
Wim Taymans
6a5293d065 gst-libs/gst/audio/gstringbuffer.c: Don't try to call the delay method when the device is not opened.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_delay):
Don't try to call the delay method when the device is not
opened.
2005-05-31 11:38:10 +00:00
Wim Taymans
5474600d4f gst-libs/gst/audio/: Various small cleanups.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c:
(gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init),
(audioringbuffer_thread_func), (gst_audioringbuffer_init),
(gst_audioringbuffer_acquire), (gst_audioringbuffer_release),
(gst_audioringbuffer_play), (gst_audioringbuffer_stop),
(gst_audioringbuffer_delay), (gst_audiosink_class_init),
(gst_audiosink_create_ringbuffer):
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_baseaudiosink_class_init), (gst_baseaudiosink_init),
(gst_baseaudiosink_get_clock), (gst_baseaudiosink_get_time),
(gst_baseaudiosink_set_property), (build_linear_format),
(debug_spec_caps), (debug_spec_buffer),
(gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times),
(gst_baseaudiosink_event), (gst_baseaudiosink_preroll),
(gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer),
(gst_baseaudiosink_callback), (gst_baseaudiosink_change_state):
* gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type),
(gst_ringbuffer_set_callback), (gst_ringbuffer_acquire),
(gst_ringbuffer_release), (gst_ringbuffer_is_acquired),
(gst_ringbuffer_play), (gst_ringbuffer_pause),
(gst_ringbuffer_stop), (gst_ringbuffer_delay),
(gst_ringbuffer_played_samples), (gst_ringbuffer_set_sample),
(wait_segment), (gst_ringbuffer_commit),
(gst_ringbuffer_prepare_read), (gst_ringbuffer_advance),
(gst_ringbuffer_clear):
Various small cleanups.

* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain),
(gst_audio_convert_change_state):
* gst/subparse/gstsubparse.c: (gst_subparse_chain):
No need to take the locks anymore.
2005-05-25 19:52:14 +00:00
Ronald S. Bultje
a159660dfc gst-libs/gst/audio/gstringbuffer.c: This can't be good.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_clear):
This can't be good.
2005-05-23 18:07:28 +00:00
Wim Taymans
62c46c1480 gst-libs/gst/net/: Added buffer subclass to store extra to/from addresses for network sources/sinks.
Original commit message from CVS:
* gst-libs/gst/net/Makefile.am:
* gst-libs/gst/net/README:
* gst-libs/gst/net/gstnetbuffer.c: (gst_netbuffer_get_type),
(gst_netbuffer_class_init), (gst_netbuffer_init),
(gst_netbuffer_finalize), (gst_netbuffer_copy),
(gst_netbuffer_new), (gst_netaddress_set_ip4_address),
(gst_netaddress_set_ip6_address), (gst_netaddress_get_net_type),
(gst_netaddress_get_ip4_address), (gst_netaddress_get_ip6_address):
* gst-libs/gst/net/gstnetbuffer.h:
Added buffer subclass to store extra to/from addresses for
network sources/sinks.
2005-05-19 11:56:48 +00:00
Ronald S. Bultje
8326cc5387 gst-libs/gst/gconf/gconf.c: Don't lock an unassigned variable.
Original commit message from CVS:
* gst-libs/gst/gconf/gconf.c: (gst_bin_find_unconnected_pad):
Don't lock an unassigned variable.
2005-05-18 15:54:00 +00:00
Wim Taymans
9fccefe949 gst/: Fix passthrough in ffmpegcolorspace.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiofilter.c: (gst_audiofilter_link),
(gst_audiofilter_chain):
* gst-libs/gst/audio/gstaudiosink.c:
(gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init),
(audioringbuffer_thread_func), (gst_audioringbuffer_init),
(gst_audioringbuffer_acquire), (gst_audioringbuffer_release),
(gst_audioringbuffer_play), (gst_audioringbuffer_stop),
(gst_audioringbuffer_delay), (gst_audiosink_class_init),
(gst_audiosink_create_ringbuffer):
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain),
(gst_audio_convert_caps_remove_format_info),
(gst_audio_convert_getcaps), (gst_audio_convert_setcaps),
(gst_audio_convert_fixate), (gst_audio_convert_channels):
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_getcaps), (gst_ffmpegcsp_configure_context),
(gst_ffmpegcsp_setcaps), (gst_ffmpegcsp_chain):
Fix passthrough in ffmpegcolorspace.
Fix memset in audiosink on wrong memory.
2005-05-17 10:47:02 +00:00
David Schleef
d90ee5bfa3 Port from GstData to GstMiniObject.
Original commit message from CVS:
Port from GstData to GstMiniObject.
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_dispose):
* ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page),
(gst_ogg_mux_queue_pads), (gst_ogg_mux_set_header_on_caps),
(gst_ogg_mux_collected):
* ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
* ext/theora/theoradec.c: (theora_handle_comment_packet),
(theora_handle_data_packet):
* ext/theora/theoraenc.c: (theora_buffer_from_packet),
(theora_set_header_on_caps), (theora_enc_chain):
* ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
(vorbis_handle_comment_packet):
* ext/vorbis/vorbisenc.c: (gst_vorbisenc_set_header_on_caps):
* ext/vorbis/vorbisparse.c: (vorbis_parse_set_header_on_caps):
* gst-libs/gst/audio/gstaudiofilter.c: (gst_audiofilter_chain):
* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_chain):
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_get_buffer):
* gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
* gst/playback/gstplaybasebin.c: (check_queue), (probe_triggered),
(mute_stream), (silence_stream):
* gst/playback/gstplaybin.c: (gst_play_bin_class_init):
* gst/volume/gstvolume.c: (volume_transform):
* sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
(gst_ximage_buffer_init), (gst_ximage_buffer_class_init),
(gst_ximage_buffer_get_type), (gst_ximagesink_check_xshm_calls),
(gst_ximagesink_ximage_new), (gst_ximagesink_ximage_destroy),
(gst_ximagesink_ximage_put), (gst_ximagesink_imagepool_clear),
(gst_ximagesink_show_frame), (gst_ximagesink_buffer_free),
(gst_ximagesink_buffer_alloc):
* sys/ximage/ximagesink.h:
2005-05-16 15:35:52 +00:00
Jan Schmidt
8a124c2c66 configure.ac: Disable libvisual
Original commit message from CVS:
* configure.ac:
Disable libvisual

* examples/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/video/Makefile.am:
Fixups for missing variables.
2005-05-09 11:55:12 +00:00
Wim Taymans
fa8c2eb659 Make the base audiosink return an error when there is no audiobuffer negotiated.
Original commit message from CVS:
Make the base audiosink return an error when there is no
audiobuffer negotiated.
2005-05-06 16:18:24 +00:00
Christian Schaller
393df2e486 add ported videofilter to cvs head
Original commit message from CVS:
add ported videofilter to cvs head
2005-05-06 11:25:56 +00:00
Andy Wingo
790c059867 gst/: Some GCC4 fixes
Original commit message from CVS:
2005-05-05  Andy Wingo  <wingo@pobox.com>

* gst-libs/gst/tag/gstid3tag.c: (gst_tag_list_new_from_id3v1):
* gst-libs/gst/tag/gstvorbistag.c:
(gst_tag_list_from_vorbiscomment_buffer), (gst_vorbis_tag_chain):
* gst/adder/gstadder.h:
* gst/audioconvert/gstchannelmix.c:
(gst_audio_convert_fill_one_other):
* gst/audiorate/gstaudiorate.c: (gst_audiorate_setcaps),
(gst_audiorate_init), (gst_audiorate_chain):
* gst/playback/gstplaybasebin.c: (setup_source):
* gst/playback/test3.c: (update_scale):
Some GCC4 fixes

* po/af.po:
* po/az.po:
* po/cs.po:
* po/en_GB.po:
* po/hu.po:
* po/it.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/uk.po:
* po/vi.po: Foo
2005-05-05 14:57:20 +00:00
Wim Taymans
b9a30899dd GCC 4 compile fixes
Original commit message from CVS:
GCC 4 compile fixes
2005-05-05 10:42:41 +00:00
Wim Taymans
658bd2cac6 More work on the audiosink, mostly debugging and a race in shutdown.
Original commit message from CVS:
* docs/design-audiosinks.txt:
* gst-libs/gst/audio/TODO:
* gst-libs/gst/audio/gstaudiosink.c:
(gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init),
(audioringbuffer_thread_func), (gst_audioringbuffer_init),
(gst_audioringbuffer_acquire), (gst_audioringbuffer_release),
(gst_audioringbuffer_play), (gst_audioringbuffer_stop),
(gst_audioringbuffer_delay), (gst_audiosink_class_init),
(gst_audiosink_create_ringbuffer):
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_baseaudiosink_class_init), (gst_baseaudiosink_init),
(gst_baseaudiosink_get_clock), (gst_baseaudiosink_get_time),
(gst_baseaudiosink_set_property), (build_linear_format),
(debug_spec_caps), (debug_spec_buffer),
(gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times),
(gst_baseaudiosink_event), (gst_baseaudiosink_preroll),
(gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer),
(gst_baseaudiosink_callback), (gst_baseaudiosink_change_state):
* gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type),
(gst_ringbuffer_set_callback), (gst_ringbuffer_acquire),
(gst_ringbuffer_release), (gst_ringbuffer_play),
(gst_ringbuffer_pause), (gst_ringbuffer_stop),
(gst_ringbuffer_delay), (gst_ringbuffer_played_samples),
(gst_ringbuffer_set_sample), (wait_segment),
(gst_ringbuffer_commit), (gst_ringbuffer_prepare_read),
(gst_ringbuffer_advance), (gst_ringbuffer_clear):
More work on the audiosink, mostly debugging and a race in
shutdown.
2005-05-05 09:37:46 +00:00
Wim Taymans
235ea5989c Make ringbuffer faster and more simple by removing the locks in the playback thread.
Original commit message from CVS:
Make ringbuffer faster and more simple by removing the locks
in the playback thread.
Add sample accurate playback based on buffer sample offsets.
Make the baseaudiosink provide a clock.
Parse caps in the base class.
Correctly handle seeking, flushing and state changes.
2005-04-28 16:15:42 +00:00
Ronald S. Bultje
8664d3ff31 Remove media-info, which is also successed by playbin (see Totem implementation).
Original commit message from CVS:
* configure.ac:
* gst-libs/gst/Makefile.am:
* gst-libs/gst/media-info/.cvsignore:
* gst-libs/gst/media-info/Makefile.am:
* gst-libs/gst/media-info/README:
* gst-libs/gst/media-info/media-info-priv.c:
* gst-libs/gst/media-info/media-info-priv.h:
* gst-libs/gst/media-info/media-info-test.c:
* gst-libs/gst/media-info/media-info.c:
* gst-libs/gst/media-info/media-info.h:
* gst-libs/gst/media-info/media-info.vcproj:
* pkgconfig/Makefile.am:
* pkgconfig/gstreamer-media-info-uninstalled.pc.in:
* pkgconfig/gstreamer-media-info.pc.in:
Remove media-info, which is also successed by playbin (see Totem
implementation).
2005-04-25 10:15:12 +00:00
Ronald S. Bultje
7172a58563 Remove libgstplay, playbin is now the official successor.
Original commit message from CVS:
* configure.ac:
* examples/Makefile.am:
* examples/gstplay/.cvsignore:
* examples/gstplay/Makefile.am:
* examples/gstplay/player.c:
* gst-libs/gst/Makefile.am:
* gst-libs/gst/play/.cvsignore:
* gst-libs/gst/play/Makefile.am:
* gst-libs/gst/play/play.c:
* gst-libs/gst/play/play.h:
* gst-libs/gst/play/play.vcproj:
* pkgconfig/Makefile.am:
* pkgconfig/gstreamer-play-uninstalled.pc.in:
* pkgconfig/gstreamer-play.pc.in:
Remove libgstplay, playbin is now the official successor.
2005-04-25 09:33:35 +00:00
Ronald S. Bultje
3420e480bc Remove deprecated xwindowlistener (I've moved xwindowlistening in the v4l/v4l2 plugins over to serverside).
Original commit message from CVS:
* configure.ac:
* gst-libs/gst/Makefile.am:
* gst-libs/gst/xwindowlistener/Makefile.am:
* gst-libs/gst/xwindowlistener/xwindowlistener.c:
* gst-libs/gst/xwindowlistener/xwindowlistener.h:
Remove deprecated xwindowlistener (I've moved xwindowlistening
in the v4l/v4l2 plugins over to serverside).
2005-04-25 08:53:41 +00:00
David Schleef
ab06cc8f10 Don't use GST_PLUGIN_LDFLAGS, because these aren't plugins.
Original commit message from CVS:
Don't use GST_PLUGIN_LDFLAGS, because these aren't plugins.
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/video/Makefile.am:
* gst-libs/gst/xwindowlistener/Makefile.am:
Convert to 0.9 API, seems to work:
* sys/ximage/Makefile.am:
* sys/ximage/ximagesink.c:
2005-04-25 07:06:09 +00:00
David Schleef
129c7e8af1 configure.ac: Remove idct and resample libs
Original commit message from CVS:
* configure.ac: Remove idct and resample libs
* gst-libs/gst/Makefile.am: same
Remove usage of gst_library_load():
* ext/alsa/gstalsaplugin.c: (plugin_init):
* ext/libvisual/visual.c: (plugin_init):
* ext/ogg/gstogg.c: (plugin_init):
* ext/theora/theora.c: (plugin_init):
* ext/vorbis/vorbis.c: (plugin_init):
* gst-libs/gst/audio/gstaudiofiltertemplate.c: (plugin_init):
* gst/audioscale/gstaudioscale.c:
* gst/adder/gstadder.c: (plugin_init):
* gst/audioconvert/plugin.c: (plugin_init):
* sys/ximage/ximagesink.c: (plugin_init):
* sys/xvimage/xvimagesink.c: (plugin_init):
* gst/tcp/gsttcpplugin.c: (plugin_init):
Link plugins against libraries:
* ext/ogg/Makefile.am:
* ext/theora/Makefile.am:
* ext/vorbis/Makefile.am:
* gst/audioconvert/Makefile.am:
Create proper libraries:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/video/Makefile.am:
Move resample library to audioscale plugin directory:
* gst-libs/gst/resample/Makefile.am:
* gst-libs/gst/resample/README:
* gst-libs/gst/resample/dtof.c:
* gst-libs/gst/resample/dtos.c:
* gst-libs/gst/resample/functable.c:
* gst-libs/gst/resample/private.h:
* gst-libs/gst/resample/resample.c:
* gst-libs/gst/resample/resample.h:
* gst-libs/gst/resample/resample.vcproj:
* gst-libs/gst/resample/test.c:
* gst/audioscale/Makefile.am:
* gst/audioscale/README:
* gst/audioscale/dtof.c:
* gst/audioscale/dtos.c:
* gst/audioscale/functable.c:
* gst/audioscale/private.h:
* gst/audioscale/resample.c:
* gst/audioscale/resample.h:
* gst/audioscale/test.c:
Move tagedit library to gst-libs:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/tag/gstid3tag.c:
* gst-libs/gst/tag/gsttagediting.c:
* gst-libs/gst/tag/gsttageditingprivate.h:
* gst-libs/gst/tag/gstvorbistag.c:
* gst/tags/Makefile.am:
* gst/tags/gstid3tag.c:
* gst/tags/gstvorbistag.c:
Fix for core changes:
* gst/sine/gstsinesrc.c: (gst_sinesrc_class_init),
(gst_sinesrc_init), (gst_sinesrc_src_fixate), (gst_sinesrc_link),
(gst_sinesrc_getrange):
2005-04-25 00:23:06 +00:00
David Schleef
f6e65158b0 gst-libs/gst/Makefile.am: Remove idct. It hasn't been used in gst-plugins in a long time, and properly belongs in li...
Original commit message from CVS:
* gst-libs/gst/Makefile.am: Remove idct.  It hasn't been used
in gst-plugins in a long time, and properly belongs in liboil.
* gst-libs/gst/idct/Makefile.am:
* gst-libs/gst/idct/README:
* gst-libs/gst/idct/dct.h:
* gst-libs/gst/idct/doieee:
* gst-libs/gst/idct/fastintidct.c:
* gst-libs/gst/idct/floatidct.c:
* gst-libs/gst/idct/idct.c:
* gst-libs/gst/idct/idct.h:
* gst-libs/gst/idct/idtc.vcproj:
* gst-libs/gst/idct/ieeetest.c:
* gst-libs/gst/idct/intidct.c:
2005-04-23 20:59:48 +00:00
Wim Taymans
5a3941c762 An attempt at a set of audio base classes together with some design docs.
Original commit message from CVS:
* docs/design-audiosinks.txt:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/audio/TODO:
* gst-libs/gst/audio/gstaudiosink.c:
(gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init),
(audioringbuffer_thread_func), (gst_audioringbuffer_init),
(gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize),
(gst_audioringbuffer_acquire), (gst_audioringbuffer_release),
(gst_audioringbuffer_play), (gst_audioringbuffer_stop),
(gst_audioringbuffer_delay), (gst_audiosink_base_init),
(gst_audiosink_class_init), (gst_audiosink_init),
(gst_audiosink_create_ringbuffer):
* gst-libs/gst/audio/gstaudiosink.h:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init),
(gst_baseaudiosink_init), (gst_baseaudiosink_set_property),
(gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps),
(gst_baseaudiosink_get_times), (gst_baseaudiosink_event),
(gst_baseaudiosink_preroll), (gst_baseaudiosink_render),
(gst_baseaudiosink_create_ringbuffer),
(gst_baseaudiosink_callback), (gst_baseaudiosink_change_state):
* gst-libs/gst/audio/gstbaseaudiosink.h:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type),
(gst_ringbuffer_class_init), (gst_ringbuffer_init),
(gst_ringbuffer_dispose), (gst_ringbuffer_finalize),
(gst_ringbuffer_set_callback), (gst_ringbuffer_acquire),
(gst_ringbuffer_release), (gst_ringbuffer_play_unlocked),
(gst_ringbuffer_play), (gst_ringbuffer_pause),
(gst_ringbuffer_resume), (gst_ringbuffer_stop),
(gst_ringbuffer_callback), (gst_ringbuffer_delay),
(gst_ringbuffer_played_samples), (gst_ringbuffer_commit),
(gst_ringbuffer_prepare_read), (gst_ringbuffer_clear):
* gst-libs/gst/audio/gstringbuffer.h:
An attempt at a set of audio base classes together with some
design docs.
2005-04-20 10:19:54 +00:00
Wim Taymans
73d7c02993 Make gnomevfssrc extend the source base class.
Original commit message from CVS:
* ext/gnomevfs/Makefile.am:
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_get_type),
(gst_gnomevfssrc_class_init), (gst_gnomevfssrc_init),
(gst_gnomevfssrc_set_property), (gst_gnomevfssrc_get_property),
(gst_gnomevfssrc_create), (gst_gnomevfssrc_is_seekable),
(gst_gnomevfssrc_get_size), (gst_gnomevfssrc_start),
(gst_gnomevfssrc_stop):
* ext/ogg/Makefile.am:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_get_data),
(gst_ogg_demux_find_chains), (gst_ogg_demux_sink_activate):
* ext/theora/Makefile.am:
* ext/theora/theoradec.c: (_inc_granulepos),
(theora_dec_sink_event), (theora_dec_chain):
* ext/vorbis/Makefile.am:
* ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
(vorbis_dec_sink_event), (vorbis_dec_chain):
* gst-libs/gst/audio/Makefile.am:
* sys/xvimage/Makefile.am:
Make gnomevfssrc extend the source base class.
Fix linking against libs in various plugins.
2005-04-06 17:33:07 +00:00
Andy Wingo
5967f9631d gst-libs/gst/video/Makefile.am (libgstvideo_la_LDFLAGS): Use
Original commit message from CVS:
2005-04-06  Andy Wingo  <wingo@pobox.com>

* gst-libs/gst/video/Makefile.am (libgstvideo_la_LDFLAGS): Use
GST_BASE_LIBS.
2005-04-06 11:34:30 +00:00
Wim Taymans
1dae961cbf Plugin port to 0.9, ogg/theora playback should work in the seek example now.
Original commit message from CVS:
Plugin port to 0.9, ogg/theora playback should work in the seek
example now.
Removed old examples.
Removed old oggvorbisenc, renamed rawvorbisenc to vorbisenc as
explained in 0.9 TODO doc.
2005-03-31 09:43:49 +00:00
Tim-Philipp Müller
a84c1f2d5d gst-libs/gst/riff/riff-media.c: Do actually fix invalid RIFF fmt header values for alaw and mulaw audio instead of ju...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_audio_caps_with_data):
Do actually fix invalid RIFF fmt header values for alaw
and mulaw audio instead of just saying so.

* gst/wavparse/gstwavparse.c: (gst_wavparse_fmt):
Give gst_riff_create_audio_caps_with_data() a chance to
fix up broken format header fields before extracting any
parameters from the header. (fixes #167633)
2005-02-20 12:49:19 +00:00
Tim-Philipp Müller
38f505b3c8 Add G_BEGIN_DECLS and G_END_DECLS around headers where missing, so that they work when included from C++ code
Original commit message from CVS:
Add G_BEGIN_DECLS and G_END_DECLS around headers where missing, so that they work when included from C++ code
2005-02-09 22:31:05 +00:00
David Schleef
4785be863a configure.ac: Put DEFAULT_AUDIOSINK in config.h and use whereever possible. (Fixes #165997)
Original commit message from CVS:
* configure.ac: Put DEFAULT_AUDIOSINK in config.h and use
whereever possible.  (Fixes #165997)
* examples/capsfilter/capsfilter1.c: (main):
* examples/dynparams/filter.c: (create_ui):
* examples/seeking/cdparanoia.c: (get_track_info), (main):
* examples/seeking/chained.c: (main):
* examples/seeking/seek.c: (make_mod_pipeline), (make_dv_pipeline),
(make_wav_pipeline), (make_flac_pipeline), (make_sid_pipeline),
(make_vorbis_pipeline), (make_mp3_pipeline), (make_avi_pipeline),
(make_mpeg_pipeline), (make_mpegnt_pipeline):
* examples/seeking/spider_seek.c: (make_spider_pipeline):
* examples/switch/switcher.c: (main):
* ext/dv/demo-play.c: (main):
* ext/faad/gstfaad.c: (gst_faad_change_state):
* ext/mad/gstmad.c: (gst_mad_chain):
* ext/smoothwave/demo-osssrc.c: (main):
* gst-libs/gst/gconf/gconf.c: (gst_gconf_set_string),
(gst_gconf_render_bin_from_description),
(gst_gconf_get_default_audio_sink),
(gst_gconf_get_default_video_sink),
(gst_gconf_get_default_audio_src),
(gst_gconf_get_default_video_src),
(gst_gconf_get_default_visualization_element):
* gst/level/demo.c: (main):
* gst/level/plot.c: (main):
* gst/playback/gstplaybin.c: (gen_video_element),
(gen_audio_element):
* gst/playback/test.c: (gen_video_element), (gen_audio_element):
* gst/playondemand/demo-mp3.c: (setup_pipeline):
* gst/sine/demo-dparams.c: (main):
* gst/spectrum/demo-osssrc.c: (main):
* gst/speed/demo-mp3.c: (main):
* gst/volume/demo.c: (main):
* testsuite/embed/embed.c: (main):
2005-02-02 08:14:01 +00:00
Ronald S. Bultje
3e29d49da3 gst-libs/gst/riff/riff-media.c: Add extradata to huffyuv (fixes #165013).
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data):
Add extradata to huffyuv (fixes #165013).
* gst-libs/gst/riff/riff-read.c:
(gst_riff_read_strf_vids_with_data):
Fix extradata extraction if it is in the chunk size.
2005-01-25 15:17:23 +00:00
Ronald S. Bultje
e49e1cf43d gst-libs/gst/riff/riff-media.c: Audio can be <8000Hz.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_audio_caps_with_data):
Audio can be <8000Hz.
2005-01-24 18:41:18 +00:00
Ronald S. Bultje
75a10dfa53 gst-libs/gst/riff/riff-read.*: Add _peek version (req'ed in CDXA).
Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c: (gst_riff_peek_element_data),
(gst_riff_read_element_data):
* gst-libs/gst/riff/riff-read.h:
Add _peek version (req'ed in CDXA).
* gst/cdxaparse/gstcdxaparse.c: (gst_cdxaparse_init),
(gst_cdxaparse_loop):
Fix parsing in playbin.
* gst/playback/gstdecodebin.c: (close_pad_link):
Ignore current_ pads, they cause major annoyance.
2005-01-19 22:42:21 +00:00
Ronald S. Bultje
1e37668bb5 gst-libs/gst/riff/riff-media.c: Add intel-h263.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data),
(gst_riff_create_video_template_caps):
Add intel-h263.
2005-01-19 12:47:20 +00:00
Thomas Vander Stichele
c5c4fe1d26 ignore more
Original commit message from CVS:
ignore more
2005-01-17 12:45:48 +00:00
Thomas Vander Stichele
802e07ad33 ignore generated files
Original commit message from CVS:
ignore generated files
2005-01-17 12:38:17 +00:00
Ronald S. Bultje
a321095c80 gst-libs/gst/riff/riff-read.c: Don't bail on unknown events.
Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_use_event):
Don't bail on unknown events.
* gst/audioscale/gstaudioscale.c: (gst_audioscale_chain):
Don't crash on events before negotiation.
* gst/avi/gstavidemux.c: (gst_avi_demux_add_stream):
Send tags on pads, too.
* gst/playback/gststreamselector.c:
(gst_stream_selector_request_new_pad):
Forward events on first pad if no input was selected yet.
2005-01-10 16:09:25 +00:00
Ronald S. Bultje
4dced785fe Remove all references to xvideosink, fix examples (#140845).
Original commit message from CVS:
* configure.ac:
* examples/capsfilter/capsfilter1.c: (main):
* examples/seeking/spider_seek.c: (make_spider_pipeline):
* ext/dvdread/Makefile.am:
* ext/dvdread/demo-play:
* ext/dvdread/demo-play.c:
* gconf/gstreamer.schemas.in:
* gst-libs/gst/gconf/gconf.c:
* sys/v4l/TODO:
* testsuite/Makefile.am:
* testsuite/embed/Makefile.am:
* testsuite/embed/embed.c: (cb_expose), (main):
Remove all references to xvideosink, fix examples (#140845).
* gst/playback/gstplaybasebin.c: (group_destroy):
Apparently, disposal does not unlink - so do explicitely.
* ext/alsa/gstalsasink.c: (gst_alsa_sink_check_event):
Add debug.
2005-01-09 14:53:59 +00:00
Stéphane Loeuillet
dc10c09315 gst/: Add AMR (VBR and CBR) ids to riff.h audio codec list
Original commit message from CVS:
* gst-libs/gst/riff/riff-ids.h:
* gst/wavenc/riff.h:
Add AMR (VBR and CBR) ids to riff.h audio codec list

* gst/asfdemux/gstasfdemux.c:
(gst_asf_demux_process_ext_content_desc),
(gst_asf_demux_process_object):
Retrieve more tags from ASF files (Genre, AlbumTitle, Artist)
2005-01-05 21:46:19 +00:00
Ronald S. Bultje
94fed1fa06 gst-libs/gst/resample/resample.c: Fix invalid memory access (#159211).
Original commit message from CVS:
Reviewed by:  Ronald S. Bultje  <rbultje@ronald.bitfreak.net>
* gst-libs/gst/resample/resample.c: (gst_resample_sinc_ft_s16):
Fix invalid memory access (#159211).
2005-01-05 15:04:02 +00:00
Ronald S. Bultje
6e1c77c85e examples/gstplay/player.c: Don't iterate.
Original commit message from CVS:
* examples/gstplay/player.c: (main):
Don't iterate.
* examples/seeking/seek.c: (fixate), (make_playerbin_pipeline):
Add visualizations.
* ext/a52dec/gsta52dec.c: (gst_a52dec_push),
(gst_a52dec_handle_frame):
Set duration.
* ext/dvdnav/gst-dvd:
Add audioconvert. Fixes #161325.
* ext/dvdread/dvdreadsrc.c: (dvdreadsrc_get):
Explicitely case to gint64. Possible valgrind error.
* gst-libs/gst/play/play.c: (caps_set), (setup_size),
(gst_play_tick_callback), (gst_play_change_state),
(gst_play_dispose), (gst_play_init), (gst_play_class_init),
(gst_play_set_location), (gst_play_get_location),
(gst_play_seek_to_time), (gst_play_set_data_src),
(gst_play_set_video_sink), (gst_play_set_audio_sink),
(gst_play_set_visualization), (gst_play_connect_visualization),
(gst_play_get_framerate), (gst_play_get_all_by_interface),
(gst_play_new):
Use playbin. Fixes #139749 and #147744.
* gst/apetag/apedemux.c: (gst_ape_demux_parse_tags):
Add genre tag.
* gst/audioscale/gstaudioscale.c: (gst_audioscale_method_get_type),
(audioscale_get_type), (gst_audioscale_base_init),
(gst_audioscale_class_init), (gst_audioscale_expand_caps),
(gst_audioscale_getcaps), (gst_audioscale_fixate),
(gst_audioscale_link), (gst_audioscale_get_buffer),
(gst_audioscale_decrease_rate), (gst_audioscale_increase_rate),
(gst_audioscale_init), (gst_audioscale_dispose),
(gst_audioscale_chain), (gst_audioscale_set_property),
(gst_audioscale_get_property), (plugin_init):
Indent properly.
* gst/mpegstream/gstdvddemux.c: (gst_dvd_demux_process_private):
Fix LPCM.
* gst/qtdemux/qtdemux.c: (qtdemux_parse_udta),
(qtdemux_tag_add_str), (qtdemux_tag_add_num),
(qtdemux_tag_add_gnre), (qtdemux_video_caps):
Add more metadata (fixes #162656).
2005-01-05 14:56:27 +00:00
Ronald S. Bultje
50f80de3be gst-libs/gst/riff/riff-media.c: Add BLZ0 (Blizzard's version of DivX) fourcc.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data):
Add BLZ0 (Blizzard's version of DivX) fourcc.
2004-12-19 11:10:32 +00:00
Ronald S. Bultje
1b0cfd03d5 gst-libs/gst/riff/riff-read.c: Read extradata correctly (fixes #155879).
Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c:
(gst_riff_read_strf_auds_with_data):
Read extradata correctly (fixes #155879).
2004-12-16 23:57:26 +00:00
Ronald S. Bultje
95adb1fdd8 gst-libs/gst/riff/riff-media.c: Add h264.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data),
(gst_riff_create_video_template_caps):
Add h264.
2004-12-16 20:30:00 +00:00
Ronald S. Bultje
405f640d2a gst-libs/gst/audio/Makefile.am: Try to fix buildbot.
Original commit message from CVS:
* gst-libs/gst/audio/Makefile.am:
Try to fix buildbot.
2004-12-16 19:45:32 +00:00
Ronald S. Bultje
5ab122366b gst/: Fix memleak (#159215).
Original commit message from CVS:
Reviewed by:  Ronald S. Bultje  <rbultje@ronald.bitfreak.net>
* gst-libs/gst/resample/resample.c: (gst_resample_close):
* gst-libs/gst/resample/resample.h:
* gst/audioscale/gstaudioscale.c:
Fix memleak (#159215).
2004-12-16 11:39:00 +00:00
David Schleef
180a910236 configure.ac: add audioresample and cairo plugins. Remove
Original commit message from CVS:
* configure.ac: add audioresample and cairo plugins.  Remove
HAVE_MMX stuff, because it's not used.
* ext/Makefile.am: same
* ext/audioresample/Makefile.am: You are not ready for an
audio resampling element based on audioresample.
* ext/audioresample/gstaudioresample.c:
* ext/audioresample/gstaudioresample.h:
* ext/cairo/Makefile.am: You are not ready for overlay elements
based on cairo.  Don't look too closely, these elements kinda
suck right now.
* ext/cairo/gstcairo.c: new
* ext/cairo/gsttextoverlay.c: new
* ext/cairo/gsttextoverlay.h: new
* ext/cairo/gsttimeoverlay.c: new
* ext/cairo/gsttimeoverlay.h: new
* gst-libs/gst/media-info/media-info-priv.h: fix compile
problem with compilers that don't support variadic macros.
2004-12-16 05:32:07 +00:00
Stéphane Loeuillet
2e32ed0f5b forgot to add H264 to avidemux template caps
Original commit message from CVS:
forgot to add H264 to avidemux template caps
2004-12-13 18:09:34 +00:00
Stéphane Loeuillet
ba02b1b4f0 add VSSH (VideoSoft h264) and remove s323 (h323) from riff-lib because s323 is quicktime specific
Original commit message from CVS:
add VSSH (VideoSoft h264) and remove s323 (h323) from riff-lib
because s323 is quicktime specific
2004-12-13 12:11:57 +00:00
Stéphane Loeuillet
803f02daaa gst/asfdemux/README gst/wavenc/riff.h gst-libs/gst/riff/riff-ids.h gst-libs/gst/riff/riff-media.c add new 4CC codes f...
Original commit message from CVS:
* gst/asfdemux/README
* gst/wavenc/riff.h
* gst-libs/gst/riff/riff-ids.h
* gst-libs/gst/riff/riff-media.c
add new 4CC codes for h263 related codecs
fixes partially bug #155163
2004-12-13 00:47:20 +00:00
Ronald S. Bultje
98298a8c56 ext/faad/gstfaad.c: Set DURATION even if source buffer didn't. Also use increasing timestamps.
Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_chanpos_from_gst),
(gst_faad_chanpos_to_gst), (gst_faad_chain):
Set DURATION even if source buffer didn't. Also use increasing
timestamps.
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_audio_caps_with_data):
Block_align can have larger values than 8192.
2004-12-03 15:34:19 +00:00
Ronald S. Bultje
4fe1a737b3 ext/esd/esdsink.c: Make error actually say something useful (fixes #156798).
Original commit message from CVS:
* ext/esd/esdsink.c: (gst_esdsink_chain):
Make error actually say something useful (fixes #156798).
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data),
(gst_riff_create_video_template_caps):
Add Intel Video 5.0 fourcc (IV50).
2004-12-01 20:42:01 +00:00
Ronald S. Bultje
5e10a662b3 gst-libs/gst/riff/riff-read.c: Don't forward DISCONT events (fixes #159684).
Original commit message from CVS:
Reviewed by:  Ronald S. Bultje  <rbultje@ronald.bitfreak.net>
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_use_event):
Don't forward DISCONT events (fixes #159684).
2004-12-01 13:23:39 +00:00
Martin Soto
8ed05be101 gst-libs/gst/audio/audioclock.c (gst_audio_clock_set_active)
Original commit message from CVS:
2004-11-27  Martin Soto  <martinsoto@users.sourceforge.net>

* gst-libs/gst/audio/audioclock.c (gst_audio_clock_set_active)
(gst_audio_clock_get_internal_time):
Fix active <-> inactive transitions: ensure time value always
grows and avoid abrupt value changes.
2004-11-27 09:37:20 +00:00
Ronald S. Bultje
3a0a2898af Surround sound support.
Original commit message from CVS:
* ext/a52dec/gsta52dec.c: (gst_a52dec_channels), (gst_a52dec_push),
(gst_a52dec_reneg), (gst_a52dec_loop), (plugin_init):
* ext/alsa/gstalsa.c: (gst_alsa_get_caps):
* ext/alsa/gstalsaplugin.c: (plugin_init):
* ext/dts/gstdtsdec.c: (gst_dtsdec_channels),
(gst_dtsdec_renegotiate), (gst_dtsdec_loop), (plugin_init):
* ext/faad/gstfaad.c: (gst_faad_init), (gst_faad_chanpos_from_gst),
(gst_faad_chanpos_to_gst), (gst_faad_sinkconnect),
(gst_faad_srcgetcaps), (gst_faad_srcconnect), (gst_faad_chain),
(gst_faad_change_state), (plugin_init):
* ext/faad/gstfaad.h:
* ext/vorbis/vorbis.c: (plugin_init):
* ext/vorbis/vorbisdec.c: (vorbis_dec_chain):
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/audio/audio.c: (plugin_init):
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions),
(gst_audio_get_channel_positions),
(gst_audio_set_channel_positions),
(gst_audio_set_structure_channel_positions_list),
(add_list_to_struct), (gst_audio_set_caps_channel_positions_list),
(gst_audio_fixate_channel_positions):
* gst-libs/gst/audio/multichannel.h:
* gst-libs/gst/audio/testchannels.c: (main):
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_class_init), (gst_audio_convert_init),
(gst_audio_convert_dispose), (gst_audio_convert_getcaps),
(gst_audio_convert_parse_caps), (gst_audio_convert_link),
(gst_audio_convert_fixate), (gst_audio_convert_channels):
* gst/audioconvert/plugin.c: (plugin_init):
Surround sound support.
2004-11-25 20:36:29 +00:00
Johan Dahlin
f0f4be7e0a Fix another typo in doc string :)
Original commit message from CVS:
Fix another typo in doc string :)
2004-11-25 12:54:24 +00:00
Johan Dahlin
9725bde877 Fix typo in doc string
Original commit message from CVS:
Fix typo in doc string
2004-11-25 12:52:24 +00:00
Benjamin Otte
37af33bdda ext/alsa/gstalsa.c: buffer-frames property was missing
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_get_caps_internal):
buffer-frames property was missing
* ext/arts/gst_arts.c:
rate missing from sinkcaps
* ext/audiofile/gstafparse.c:
* ext/audiofile/gstafsink.c:
* ext/audiofile/gstafsrc.c:
* ext/swfdec/gstswfdec.c:
int audio doesn't know buffer-frames
* ext/cdparanoia/gstcdparanoia.c:
int audio doesn't know chunksize either
* ext/nas/nassink.c:
it's endianness, not endianess
* gst-libs/gst/audio/audio.h:
make float standard pad template caps really describe float
* gst/law/mulaw.c: (linear_factory):
signed only, please
* gst/mpegstream/gstdvddemux.c:
widths of 20 are not valid
2004-11-09 06:08:22 +00:00
Jan Schmidt
4cf67a0834 Fixes a bunch of problems with finalize and dispose functions, either assumptions that dispose is only called once, o...
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_class_init), (gst_alsa_dispose),
(gst_alsa_finalize):
* ext/cdaudio/gstcdaudio.c: (gst_cdaudio_class_init),
(gst_cdaudio_finalize):
* ext/cdparanoia/gstcdparanoia.c: (cdparanoia_class_init),
(cdparanoia_finalize):
* ext/divx/gstdivxdec.c: (gst_divxdec_dispose):
* ext/divx/gstdivxenc.c: (gst_divxenc_dispose):
* ext/dvdread/dvdreadsrc.c: (dvdreadsrc_class_init),
(dvdreadsrc_finalize):
* ext/flac/gstflacdec.c: (gst_flacdec_class_init),
(gst_flacdec_finalize):
* ext/flac/gstflacenc.c: (gst_flacenc_class_init),
(gst_flacenc_finalize):
* ext/gnomevfs/gstgnomevfssink.c: (gst_gnomevfssink_class_init),
(gst_gnomevfssink_finalize):
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_class_init),
(gst_gnomevfssrc_finalize):
* ext/libfame/gstlibfame.c: (gst_fameenc_class_init),
(gst_fameenc_finalize):
* ext/nas/nassink.c: (gst_nassink_class_init),
(gst_nassink_finalize):
* ext/sdl/sdlvideosink.c: (gst_sdlvideosink_finalize),
(gst_sdlvideosink_class_init):
* ext/sndfile/gstsf.c: (gst_sf_dispose):
* gst-libs/gst/mixer/mixertrack.c: (gst_mixer_track_dispose):
* gst-libs/gst/tuner/tunerchannel.c: (gst_tuner_channel_dispose):
* gst-libs/gst/tuner/tunernorm.c: (gst_tuner_norm_dispose):
* gst-libs/gst/xwindowlistener/xwindowlistener.c:
(gst_x_window_listener_dispose):
* gst/audioscale/gstaudioscale.c:
* gst/playondemand/gstplayondemand.c: (play_on_demand_class_init),
(play_on_demand_finalize):
* gst/videofilter/gstvideobalance.c: (gst_videobalance_dispose):
* gst/videoscale/gstvideoscale.c: (gst_videoscale_chain):
* sys/cdrom/gstcdplayer.c: (cdplayer_class_init),
(cdplayer_finalize):
* sys/glsink/glimagesink.c: (gst_glimagesink_finalize),
(gst_glimagesink_class_init):
* sys/oss/gstosselement.c: (gst_osselement_class_init),
(gst_osselement_finalize):
* sys/oss/gstosssink.c: (gst_osssink_dispose):
* sys/oss/gstosssrc.c: (gst_osssrc_dispose):
* sys/v4l/gstv4lelement.c: (gst_v4lelement_dispose):
Fixes a bunch of problems with finalize and dispose functions,
either assumptions that dispose is only called once, or not calling
the parent class dispose/finalize function
2004-11-01 14:43:38 +00:00
Jan Schmidt
37ba745ec3 ext/dv/gstdvdec.c: Set EOS on the element when processing an EOS event.
Original commit message from CVS:
* ext/dv/gstdvdec.c: (gst_dvdec_handle_sink_event):
Set EOS on the element when processing an EOS event.
* ext/speex/gstspeexdec.h:
* ext/speex/gstspeexenc.h:
Only keep a const ptr to the mode
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_audio_caps_with_data),
(gst_riff_create_audio_template_caps):
Allow WMAV3, with up to 6 channels.
* gst/asfdemux/gstasfmux.c: (gst_asfmux_request_new_pad):
Don't call gst_pad_set_event_function on a sink pad.
* gst/mpegstream/gstdvddemux.c:
(gst_dvd_demux_get_subpicture_stream),
(gst_dvd_demux_set_cur_audio), (gst_dvd_demux_set_cur_subpicture):
Copy the explicit caps that were set across to the cur_* pads,
instead of trying to use a possibly non-existent negotiated caps.
Reset the type of subpicture pads to UNKNOWN after calling init_stream,
so that the caps get set.
2004-10-30 06:44:03 +00:00
Zaheer Abbas Merali
1dd50473f7 gst-libs/gst/audio/gstaudiofilter.c: fix build
Original commit message from CVS:
2004-10-28  Zaheer Abbas Merali  <zaheerabbas at merali dot org>

* gst-libs/gst/audio/gstaudiofilter.c: (gst_audiofilter_link):
fix build
2004-10-28 12:51:23 +00:00
Benjamin Otte
4f9bfe73fd gst-libs/gst/audio/gstaudiofilter.c: fix link function to always query channels and query width for floats
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiofilter.c: (gst_audiofilter_link):
fix link function to always query channels and query width for
floats
* configure.ac:
add equalizer dir
* gst/equalizer/Makefile.am:
* gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_get_type),
(gst_iir_equalizer_base_init), (gst_iir_equalizer_class_init),
(gst_iir_equalizer_init), (gst_iir_equalizer_finalize),
(arg_to_scale), (setup_filter),
(gst_iir_equalizer_compute_frequencies),
(gst_iir_equalizer_set_property), (gst_iir_equalizer_get_property),
(gst_iir_equalizer_filter_inplace), (gst_iir_equalizer_setup),
(plugin_init):
add an equalizer
2004-10-28 02:33:47 +00:00
Ronald S. Bultje
17906c7f25 gst-libs/gst/riff/riff-media.c: WMV3 missing in template caps.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_template_caps):
WMV3 missing in template caps.
2004-10-09 19:55:30 +00:00
Stéphane Loeuillet
18e3bf9591 gst/wavparse/gstwavparse.c: add ATRAC3 to STATIC CAPS to fix a warning
Original commit message from CVS:
* gst/wavparse/gstwavparse.c:
add ATRAC3 to STATIC CAPS to fix a warning

* gst/matroska/ebml-read.c:
* gst-libs/gst/riff/riff-read.c:
fix typos
2004-10-06 15:12:08 +00:00
Stéphane Loeuillet
2ff1cbabe5 gst-libs/gst/riff/riff-media.c: generate caps for ATRAC3 audio streams
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
generate caps for ATRAC3 audio streams

* gst/realmedia/rmdemux.c:
generate caps for ATRAC3 audio streams
2004-10-06 14:18:13 +00:00
Stéphane Loeuillet
1b596c9b15 gst/wavparse/gstwavparse.h remove duplicated defines for audio codec codes gst-libs/gst/riff/riff-ids.h
Original commit message from CVS:
* gst/wavparse/gstwavparse.h
remove duplicated defines for audio codec codes

* gst-libs/gst/riff/riff-ids.h
* gst/wavenc/riff.h:
add "4CC" code for ATRAC3 audio streams
add "4CC" code for ITU_G721_ADPCM (unused for now)
2004-10-06 14:11:56 +00:00
Ronald S. Bultje
5ada99461d gst-libs/gst/riff/riff-media.c: Add wing commander format mimetype/fourccs.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data),
(gst_riff_create_video_template_caps):
Add wing commander format mimetype/fourccs.
* gst/avi/gstavidemux.c: (gst_avi_demux_massage_index):
Don't crash if some value is 0.
2004-10-02 20:27:03 +00:00
Ronald S. Bultje
a9619b37f2 gst-libs/gst/riff/riff-media.c: Add DIB fourcc (raw, palettized 8-bit RGB).
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data),
(gst_riff_create_video_template_caps):
Add DIB fourcc (raw, palettized 8-bit RGB).
* gst-libs/gst/riff/riff-read.c:
(gst_riff_read_strf_vids_with_data):
Oops, fix strf_data reading bug.
* gst/avi/gstavidemux.c: (gst_avi_demux_add_stream):
Use a non-NULL tag.
* gst/qtdemux/qtdemux.c: (qtdemux_parse_trak):
Time for hacks. Sorry Dave. At least one quicktime movie (a
trailer) that I've encountered contains multiple video tracks.
One of those is the actual video track, the other are one-frame
tracks (images). Unfortunately, the number of frames according
to the trak header is 1 for each, so that doesn't help. So
instead, I look at the duration and discard tracks with a
duration shorter than 20% of the length of the stream. Better
than nothing.
2004-10-02 14:10:19 +00:00
Ronald S. Bultje
6016a9a033 gst-libs/gst/riff/riff-media.c: Add codec_data handling (like asfdemux used to do).
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data),
(gst_riff_create_audio_caps_with_data):
Add codec_data handling (like asfdemux used to do).
* gst/asfdemux/gstasf.c: (plugin_init):
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_base_init),
(gst_asf_demux_add_audio_stream), (gst_asf_demux_add_video_stream):
Use riff-media for caps creation instead of our own (mostly
broken) copy of its functions.
2004-10-01 10:50:57 +00:00
Ronald S. Bultje
eb13b061c6 ext/dvdread/dvdreadsrc.c: Fix. Don't do one big huge loop around the whole DVD, that will cache all data and thus eat...
Original commit message from CVS:
* ext/dvdread/dvdreadsrc.c: (dvdreadsrc_class_init),
(dvdreadsrc_init), (dvdreadsrc_dispose), (dvdreadsrc_set_property),
(dvdreadsrc_get_property), (_open), (_seek), (_read),
(dvdreadsrc_get), (dvdreadsrc_open_file),
(dvdreadsrc_change_state):
Fix. Don't do one big huge loop around the whole DVD, that will
cache all data and thus eat sizeof(dvd) (several GB) before we
see something.
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_seek):
Actually NULL'ify event after using it.
* gst/matroska/ebml-read.c: (gst_ebml_read_use_event),
(gst_ebml_read_handle_event), (gst_ebml_read_element_id),
(gst_ebml_read_element_length), (gst_ebml_read_element_data),
(gst_ebml_read_seek), (gst_ebml_read_skip):
Handle events.
* gst/mpegstream/gstdvddemux.c: (gst_dvd_demux_base_init),
(gst_dvd_demux_init), (gst_dvd_demux_get_audio_stream),
(gst_dvd_demux_get_subpicture_stream), (gst_dvd_demux_plugin_init):
Fix timing (this will probably break if I seek using menus, but
I didn't get there yet). VOBs and normal DVDs should now work.
Add a mpeg2-only pad with high rank so this get autoplugged for
MPEG-2 movies.
* gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_base_init),
(gst_mpeg_demux_class_init), (gst_mpeg_demux_init),
(gst_mpeg_demux_new_output_pad), (gst_mpeg_demux_get_video_stream),
(gst_mpeg_demux_get_audio_stream),
(gst_mpeg_demux_get_private_stream), (gst_mpeg_demux_parse_packet),
(gst_mpeg_demux_parse_pes), (gst_mpeg_demux_plugin_init):
Use this as second rank for MPEG-1 and MPEG-2. Still use this for
MPEG-1 but use dvddemux for MPEG-2.
* gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_class_init),
(gst_mpeg_parse_init), (gst_mpeg_parse_new_pad),
(gst_mpeg_parse_parse_packhead):
Timing. Only add pad template if it exists. Add sink template from
class and not from ourselves. This means we will always use the
correct sink template even if it is not the one defined in this
file.
2004-10-01 08:42:56 +00:00
Ronald S. Bultje
6fd179b578 ext/flac/gstflacdec.c: Only return true if we actually filled something in. Prevents player applications from showing...
Original commit message from CVS:
* ext/flac/gstflacdec.c: (gst_flacdec_src_query):
Only return true if we actually filled something in. Prevents
player applications from showing a random length for flac files.
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_class_init),
(gst_riff_read_use_event), (gst_riff_read_handle_event),
(gst_riff_read_seek), (gst_riff_read_skip), (gst_riff_read_strh),
(gst_riff_read_strf_vids_with_data),
(gst_riff_read_strf_auds_with_data), (gst_riff_read_strf_iavs):
OK, ok, so I implemented event handling. Apparently it's normal
that we receive random events at random points without asking
for it.
* gst/avi/gstavidemux.c: (gst_avi_demux_reset),
(gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query),
(gst_avi_demux_handle_src_event), (gst_avi_demux_stream_index),
(gst_avi_demux_sync), (gst_avi_demux_stream_scan),
(gst_avi_demux_massage_index), (gst_avi_demux_stream_header),
(gst_avi_demux_handle_seek), (gst_avi_demux_process_next_entry),
(gst_avi_demux_stream_data), (gst_avi_demux_loop):
* gst/avi/gstavidemux.h:
Implement non-lineair chunk handling and subchunk processing.
The first solves playback of AVI files where the audio and video
data of individual buffers that we read are not synchronized.
This should not happen according to the wonderful AVI specs, but
of course it does happen in reality. It is also a prerequisite for
the second. Subchunk processing allows us to cut chunks in small
pieces and process each of these pieces separately. This is
required because I've seen several AVI files with incredibly large
audio chunks, even some files with only one audio chunk for the
whole file. This allows for proper playback including seeking.
This patch is supposed to fix all AVI A/V sync issues.
* gst/flx/gstflxdec.c: (gst_flxdec_class_init),
(flx_decode_chunks), (flx_decode_color), (gst_flxdec_loop):
Work.
* gst/modplug/gstmodplug.cc:
Proper return value setting for the query() function.
* gst/playback/gstplaybasebin.c: (setup_source):
Being in non-playing state (after, e.g., EOS) is not necessarily
a bad thing. Allow for that. This fixes playback of short files.
They don't actually playback fully now, because the clock already
runs. This means that small files (<500kB) with a small length
(<2sec) will still not or barely play. Other files, such as mod
or flx, will work correctly, however.
2004-09-29 09:45:40 +00:00
Ronald S. Bultje
cf543aa606 ext/dirac/: Do something. Don't actually know if this works because I don't have a demuxer yet.
Original commit message from CVS:
* ext/dirac/Makefile.am:
* ext/dirac/gstdirac.cc:
* ext/dirac/gstdiracdec.cc:
* ext/dirac/gstdiracdec.h:
Do something. Don't actually know if this works because I don't
have a demuxer yet.
* ext/gsm/gstgsmdec.c: (gst_gsmdec_getcaps):
Add channels=1 to caps returned from _getcaps().
* ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_get_type),
(gst_ogm_video_parse_get_type), (gst_ogm_audio_parse_base_init),
(gst_ogm_video_parse_base_init), (gst_ogm_parse_init),
(gst_ogm_audio_parse_init), (gst_ogm_video_parse_init),
(gst_ogm_parse_sink_convert), (gst_ogm_parse_chain),
(gst_ogm_parse_change_state):
Separate between audio/video so ogmaudioparse actually uses the
audio pad templates. Both audio and video work now, including
autoplugging. Also use sometimes-srcpad hack.
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_seek):
Handle events better. Don't hang on infinite loops.
* gst/avi/gstavidemux.c: (gst_avi_demux_class_init),
(gst_avi_demux_init), (gst_avi_demux_reset),
(gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query),
(gst_avi_demux_stream_header), (gst_avi_demux_stream_data),
(gst_avi_demux_change_state):
* gst/avi/gstavidemux.h:
Improve A/V sync. Still not perfect.
* gst/matroska/ebml-read.c: (gst_ebml_read_seek),
(gst_ebml_read_skip):
Handle events better.
* gst/qtdemux/qtdemux.c: (gst_qtdemux_handle_sink_event),
(gst_qtdemux_loop_header), (qtdemux_parse_trak),
(qtdemux_audio_caps):
Add IMA4. Improve event handling. Save offset after a seek when
the headers are at the end of the file so that we don't end up in
an infinite loop.
* gst/typefind/gsttypefindfunctions.c: (qt_type_find):
Add low-priority typefind support for files with no length.
2004-09-23 14:59:22 +00:00
Christophe Fergeau
a2159103cb gst-libs/gst/riff/riff-read.c: fix infinite loop in wavparse, fixes bug
Original commit message from CVS:
2004-08-24  Sebastien Cote  <sc5@hermes.usherb.ca>

* gst-libs/gst/riff/riff-read.c: (gst_riff_peek_head),
(gst_riff_read_element_data), (gst_riff_read_seek),
(gst_riff_read_skip): fix infinite loop in wavparse, fixes bug
#144616, patch reviewed by Ronald and committed by Christophe Fergeau
<teuf@gnome.org>
2004-08-24 18:25:02 +00:00
David Schleef
ba22eaacbe gst-libs/gst/video/videosink.h: Change copyright block to LGPL.
Original commit message from CVS:
* gst-libs/gst/video/videosink.h: Change copyright block to LGPL.
2004-08-11 21:19:13 +00:00
David Schleef
024c9f3564 ext/pango/gsttextoverlay.c: Add copyright block and fix plugin license field
Original commit message from CVS:
* ext/pango/gsttextoverlay.c: Add copyright block and fix plugin
license field
* gst-libs/gst/idct/Makefile.am: Remove mmx/sse code
* gst-libs/gst/video/gstvideosink.c: Change copyright block to
LGPL.
* gst/auparse/gstauparse.c: Fix plugin license field.
* gst/monoscope/gstmonoscope.c: Fix plugin license field.
* gst/mpeg1sys/gstmpeg1systemencode.c: Fix plugin license field.
* gst/rtp/gstrtp.c: Fix plugin license field.
2004-08-11 21:06:48 +00:00
Brian Cameron
346923d634 Finished removing GPL'ed MMX code.
Original commit message from CVS:
Finished removing GPL'ed MMX code.
2004-08-09 21:30:09 +00:00
Brian Cameron
e2cb559360 Remove GPL'ed mmx32idct.c code and supporting code, since logic in gst-plugins is not supposed to be GPL'ed. This co...
Original commit message from CVS:
Remove GPL'ed mmx32idct.c code and supporting code, since logic in gst-plugins
is not supposed to be GPL'ed.  This code provided MMX optimisations, but was
never compiled in since configure never set HAVE_LIBMMX anyway.
2004-08-09 21:21:25 +00:00
Benjamin Otte
fac2d57868 fixes for G_DISABLE_ASSERT and friends
Original commit message from CVS:
* examples/dynparams/filter.c: (ui_control_create):
* examples/gstplay/player.c: (print_tag):
* ext/alsa/gstalsa.c: (gst_alsa_request_new_pad):
* ext/gdk_pixbuf/gstgdkanimation.c:
(gst_gdk_animation_iter_may_advance):
* ext/jack/gstjack.c: (gst_jack_request_new_pad):
* ext/mad/gstid3tag.c: (gst_mad_id3_to_tag_list),
(tag_list_to_id3_tag_foreach), (gst_id3_tag_handle_event):
* ext/vorbis/oggvorbisenc.c: (gst_oggvorbisenc_get_tag_value):
* ext/vorbis/vorbisenc.c: (gst_vorbisenc_get_tag_value):
* ext/xine/xineaudiodec.c: (gst_xine_audio_dec_chain):
* gst-libs/gst/media-info/media-info-test.c: (print_tag):
* gst/sine/demo-dparams.c: (main):
* gst/tags/gstvorbistag.c: (gst_tag_to_vorbis_comments):
* testsuite/alsa/formats.c: (create_pipeline):
* testsuite/alsa/sinesrc.c: (sinesrc_force_caps), (sinesrc_get):
fixes for G_DISABLE_ASSERT and friends
* gst/typefind/gsttypefindfunctions.c: (aac_type_find),
(mp3_type_frame_length_from_header), (mp3_type_find),
(plugin_init):
require mp3 typefinding to have at least MIN_HEADERS valid headers
add typefinding for AAC adts files
2004-08-03 14:28:12 +00:00
Thomas Vander Stichele
aa2c9e19b8 don't install marshal header
Original commit message from CVS:
don't install marshal header
2004-07-30 18:18:20 +00:00
Thomas Vander Stichele
9d0344af8c don't use stupid colorspace, do use hermes, make macro, mark for translation
Original commit message from CVS:
don't use stupid colorspace, do use hermes, make macro, mark for translation
2004-07-30 13:41:55 +00:00
Thomas Vander Stichele
8917ccb951 no need to link in setup stage
Original commit message from CVS:
no need to link in setup stage
2004-07-28 16:22:53 +00:00
Steve Lhomme
508098a9b9 fix local includes and 64 bits constants
Original commit message from CVS:
fix local includes and 64 bits constants
2004-07-27 21:51:32 +00:00
Steve Lhomme
7b86915b8a more working plugins
Original commit message from CVS:
more working plugins
2004-07-27 21:41:30 +00:00
Steve Lhomme
a6c379fd7a rename GStreamer-0.8.lib to libgstreamer.lib
Original commit message from CVS:
rename GStreamer-0.8.lib to libgstreamer.lib
2004-07-27 09:57:33 +00:00
Steve Lhomme
6c8a44d0c9 avoid problems with math.h, fix release dependancy
Original commit message from CVS:
avoid problems with math.h, fix release dependancy
2004-07-27 09:48:51 +00:00
Steve Lhomme
30712847fe add more plugins to the build add some definitions needed by plugins fixes for build problems
Original commit message from CVS:
add more plugins to the build
add some definitions needed by plugins
fixes for build problems
2004-07-26 22:09:59 +00:00
Steve Lhomme
6fdc736b0a more plugins supported under windows
Original commit message from CVS:
more plugins supported under windows
2004-07-26 10:03:16 +00:00
Steve Lhomme
8a934d6766 cleaned the makefiles
Original commit message from CVS:
cleaned the makefiles
2004-07-25 11:14:04 +00:00
Steve Lhomme
9be33f8db4 Copy the files where needed after building, cleaner projects
Original commit message from CVS:
Copy the files where needed after building, cleaner projects
2004-07-25 10:33:07 +00:00
Steve Lhomme
5559688599 Clean the local include
Original commit message from CVS:
Clean the local include
2004-07-25 10:28:05 +00:00
Steve Lhomme
e1d2f27a5b ok, that was not very clean
Original commit message from CVS:
ok, that was not very clean
2004-07-25 10:20:12 +00:00
Steve Lhomme
bafc38bcbf Add the preliminary canvas to build plugins on Win32
Original commit message from CVS:
Add the preliminary canvas to build plugins on Win32
2004-07-24 21:11:31 +00:00
Ronald S. Bultje
b317ec2be2 gst-libs/gst/riff/riff-media.c: Fix double end-to-native symbol conversion (#148021).
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_audio_caps_with_data):
Fix double end-to-native symbol conversion (#148021).
2004-07-20 21:06:35 +00:00
Wim Taymans
e572214cc1 gst/: Make sure we don't create 0 sized subbufers in riff-read.
Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c:
(gst_riff_read_strf_vids_with_data),
(gst_riff_read_strf_auds_with_data):
* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
(gst_avi_demux_add_stream), (gst_avi_demux_stream_header):
Make sure we don't create 0 sized subbufers in riff-read.
Signal the no more pads signal after reading the avi header.
2004-07-16 10:50:49 +00:00
Wim Taymans
4d6a0cb39e gst-libs/gst/video/video.h: Added 32 bits RGBA. Not sure if we should use another mime-type for alpha rgb. Currently ...
Original commit message from CVS:
* gst-libs/gst/video/video.h:
Added 32 bits RGBA. Not sure if we should use another mime-type
for alpha rgb. Currently the presence of the alpha_mask property
signals an alpha channel. Ronald?
2004-07-16 10:40:54 +00:00
Ronald S. Bultje
4bd27986d3 gst-libs/gst/riff/riff-media.c: mp42/mp43 (no caps) exist too.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data):
mp42/mp43 (no caps) exist too.
* gst/matroska/matroska-demux.c: (gst_matroska_demux_video_caps):
Set pixel_width/height; we've got them in-caps.
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
* gst/wavparse/gstwavparse.c: (plugin_init):
Both are valid primary.
* sys/oss/gstossmixer.c:
Remove i18n hack and enable translations.
2004-07-15 23:56:05 +00:00
Wim Taymans
1d15366460 gst/: Set codec_data on caps for avidemuxer.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_audio_caps_with_data),
(gst_riff_create_audio_caps),
(gst_riff_create_audio_template_caps):
* gst-libs/gst/riff/riff-media.h:
* gst-libs/gst/riff/riff-read.c:
(gst_riff_read_strf_vids_with_data),
(gst_riff_read_strf_auds_with_data), (gst_riff_read_strf_auds):
* gst-libs/gst/riff/riff-read.h:
* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
(gst_avi_demux_add_stream):
Set codec_data on caps for avidemuxer.
2004-07-13 10:24:29 +00:00
Wim Taymans
5b5e6c369f gst-libs/gst/riff/riff-media.c: Fix the template caps to include some more media types.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
(gst_riff_create_video_template_caps):
Fix the template caps to include some more media types.
2004-07-12 11:41:51 +00:00
Thomas Vander Stichele
7a17097201 ignore more
Original commit message from CVS:
ignore more
2004-07-12 10:13:46 +00:00
Thomas Vander Stichele
c750d9bd49 don't assert in state change
Original commit message from CVS:
don't assert in state change
2004-07-09 10:56:51 +00:00
Thomas Vander Stichele
90a779acb6 new method. various debugging
Original commit message from CVS:
new method.  various debugging
2004-07-08 17:34:08 +00:00
Thomas Vander Stichele
2391c9703d use macro to hash lookup
Original commit message from CVS:
use macro to hash lookup
2004-07-08 14:38:48 +00:00
Thomas Vander Stichele
2575f0005d gst-libs/gst/play/play.c: - add debugging info
Original commit message from CVS:

* gst-libs/gst/play/play.c: (gst_play_pipeline_setup),
(gst_play_get_length_callback), (gst_play_set_location),
(gst_play_seek_to_time), (gst_play_set_data_src),
(gst_play_set_video_sink), (gst_play_set_audio_sink),
(gst_play_set_visualization), (gst_play_connect_visualization),
(gst_play_get_sink_element):
- add debugging info
- fix looking up sink elements by iterating over complete caps
- put everything except for source and autoplugger in a complete bin
2004-07-08 13:44:06 +00:00
David Schleef
5d9d9fb859 gst-libs/gst/xoverlay/Makefile.am: xoverlay no longer depends on X. (bug #144753)
Original commit message from CVS:
* gst-libs/gst/xoverlay/Makefile.am: xoverlay no longer depends
on X.  (bug #144753)
2004-07-03 03:06:34 +00:00
Thomas Vander Stichele
0e71d88f2f clean up install/dist problem
Original commit message from CVS:
clean up install/dist problem
2004-07-02 10:02:13 +00:00
Thomas Vander Stichele
5b7573b8ff unbreak install
Original commit message from CVS:
unbreak install
2004-06-27 15:42:15 +00:00
Iain Holmes
cd0d806b11 Add name=source to the wavparse pipeline
Original commit message from CVS:
Add name=source to the wavparse pipeline
2004-06-25 11:57:25 +00:00
Benjamin Otte
814688a52b don't include -enumtypes.[ch] or -marshal.[ch] files in the disted tarball.
Original commit message from CVS:
* gst-libs/gst/colorbalance/Makefile.am:
* gst-libs/gst/mixer/Makefile.am:
* gst-libs/gst/play/Makefile.am:
* gst-libs/gst/tuner/Makefile.am:
* gst/tcp/Makefile.am:
* sys/dxr3/Makefile.am:
don't include -enumtypes.[ch] or -marshal.[ch] files in the disted
tarball.
Also add all *.list files that were missing.
* Makefile.am:
add a distcheck hook to ensure the above doesn't happen again.
2004-06-24 01:43:54 +00:00
Zaheer Abbas Merali
9ca805fb93 fixed a potential leak with previous commit
Original commit message from CVS:
2004-06-15  Zaheer Abbas Merali  <zaheerabbas at merali.org>

fixed a potential leak with previous commit

* gst-libs/gst/riff/riff-read.c: (gst_riff_peek_head):
2004-06-15 21:58:18 +00:00
Zaheer Abbas Merali
0cf31ef992 gst-libs/gst/riff/riff-read.c: Added missing refcount, fixes bug #144425
Original commit message from CVS:
2004-06-15  Zaheer Abbas Merali  <zaheerabbas at merali.org>

* gst-libs/gst/riff/riff-read.c: (gst_riff_peek_head):
Added missing refcount, fixes bug #144425
Cheers Tim for finding the bug
2004-06-15 21:25:36 +00:00
Stéphane Loeuillet
276f819c74 real fix thise time : don't use glib 2.4 specific defines
Original commit message from CVS:
real fix thise time : don't use glib 2.4 specific defines
2004-06-02 12:01:52 +00:00
Stéphane Loeuillet
23e3f829f4 zaheer :
Original commit message from CVS:
zaheer :
* gst/tcp/gsttcp.c:
* gst/tcp/gsttcpclientsrc.c:
* gst/tcp/gsttcpclientsrc.h:
* gst/tcp/gsttcpserversrc.c:
- portability fix, to compile on OSX
(fixes #143146)

* sys/osxaudio/gstosxaudioelement.c:
* sys/osxaudio/gstosxaudiosink.c:
* sys/osxaudio/gstosxaudiosrc.c:
- compilation warnings on OSX
(fixes #143153)

me :
* ext/vorbis/vorbisdec.c : sign warning fixes

* gst-libs/gst/mixer/mixertrack.c : forgoten include
to define newly used G_MAXINT32, bad owen, bad
2004-06-02 11:47:10 +00:00
Owen Fraser-Green
46616b104b Added property accessors for mixertrack and mixeroptions.
Original commit message from CVS:
Added property accessors for mixertrack and mixeroptions.
2004-06-01 20:08:52 +00:00
Thomas Vander Stichele
31e3fb26a7 more readable g_error
Original commit message from CVS:
more readable g_error
2004-06-01 08:51:55 +00:00
Stéphane Loeuillet
f47ad86b73 gst-libs/gst/tuner/tunerchannel.h: - add a freq_multiplicator field to make the conversion between internal frequency...
Original commit message from CVS:
* gst-libs/gst/tuner/tunerchannel.h:
- add a freq_multiplicator field to make the conversion
between internal frequency unit and Hz
* sys/v4l/gstv4lelement.c:
* sys/v4l2/gstv4l2element.c:
- change default video device to /dev/video0
* sys/v4l/v4l_calls.c:
* sys/v4l2/v4l2_calls.c:
- we only expose frequency to the user in Hz instead of
bastard v4lX unit (either 62.5kHz or 62.5Hz)
2004-05-31 14:16:54 +00:00
Jan Schmidt
0e141ff4ce ext/vorbis/vorbisdec.c: Initialise b_o_s and e_o_s variables
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_dec_chain):
Initialise b_o_s and e_o_s variables
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data):
Add some unusual fourcc's from mplayer avi's
* gst/multipart/multipartmux.c: (gst_multipart_mux_plugin_init):
Make the muxer have rank GST_RANK_NONE, so it doesn't mess up
autoplugging.
2004-05-31 04:56:55 +00:00
Ronald S. Bultje
8f88f9280d ext/alsa/: Add enumerations (as GstMixerOptions). Make correct distinction between input/output tracks. Add capture/p...
Original commit message from CVS:
* ext/alsa/Makefile.am:
* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_interface_init),
(gst_alsa_mixer_build_list), (gst_alsa_mixer_get_volume),
(gst_alsa_mixer_set_volume), (gst_alsa_mixer_set_mute),
(gst_alsa_mixer_set_record), (gst_alsa_mixer_set_option),
(gst_alsa_mixer_get_option):
* ext/alsa/gstalsamixer.h:
* ext/alsa/gstalsamixeroptions.c:
(gst_alsa_mixer_options_get_type),
(gst_alsa_mixer_options_class_init), (gst_alsa_mixer_options_init),
(gst_alsa_mixer_options_new):
* ext/alsa/gstalsamixeroptions.h:
* ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
* ext/alsa/gstalsamixertrack.h:
Add enumerations (as GstMixerOptions). Make correct distinction
between input/output tracks. Add capture/playback private flag.
Use flag to decide on whether to set capture or playback volumes
or switches. Use playback and record switches.
* gst-libs/gst/mixer/Makefile.am:
* gst-libs/gst/mixer/mixer-marshal.list:
* gst-libs/gst/mixer/mixer.c: (gst_mixer_class_init),
(gst_mixer_set_option), (gst_mixer_get_option),
(gst_mixer_mute_toggled), (gst_mixer_record_toggled),
(gst_mixer_volume_changed), (gst_mixer_option_changed):
* gst-libs/gst/mixer/mixer.h:
* gst-libs/gst/mixer/mixeroptions.c: (gst_mixer_options_get_type),
(gst_mixer_options_class_init), (gst_mixer_options_init),
(gst_mixer_options_dispose):
* gst-libs/gst/mixer/mixeroptions.h:
Add GstMixerOptions.
* sys/oss/gstosselement.c: (gst_osselement_class_probe_devices):
Rename Audio Mixer to OSS Mixer (similar to Alsa Mixer). Fix
broken device detection on computers with multiple OSS sound
cards.
2004-05-27 03:36:17 +00:00
Benjamin Otte
75375e26de gst-libs/gst/resample/private.h: don't use optimizations that are #if 0'ed
Original commit message from CVS:
* gst-libs/gst/resample/private.h:
don't use optimizations that are #if 0'ed
2004-05-24 16:22:12 +00:00
Stéphane Loeuillet
99ea03f95c hopefully, fix warnings in asfmux on solaris 10/with forte
Original commit message from CVS:
hopefully, fix warnings in asfmux on solaris 10/with forte
2004-05-24 14:18:56 +00:00
Stéphane Loeuillet
d8579eabe3 third batch : remove ',' at end of enums as they could confuse older gcc, foreign compilers (forte) and gtk-doc (in g...
Original commit message from CVS:
third batch :
remove ',' at end of enums as they could confuse older gcc, foreign compilers (forte) and gtk-doc
(in gst-plugins/gst-libs/ this time)
2004-05-21 23:38:49 +00:00
Stéphane Loeuillet
1f1a7cbe84 first batch : remove ',' at end of enums as they could confuse older gcc, foreign compilers (forte) and gtk-doc
Original commit message from CVS:

first batch :
remove ',' at end of enums as they could confuse older gcc, foreign compilers (forte) and gtk-doc
2004-05-21 22:39:30 +00:00
David Schleef
0ab84afae6 gst-libs/gst/resample/resample.c: Remove use of static temporary buffer. This code was obviously not supposed to las...
Original commit message from CVS:
* gst-libs/gst/resample/resample.c: (gst_resample_sinc_ft_s16),
(gst_resample_sinc_ft_float): Remove use of static temporary
buffer.  This code was obviously not supposed to last long, but
it's stuck in our ABI, so it required a little hack to make it
ABI-compatible.  Fixes #142585.
* gst-libs/gst/resample/resample.h: same.
2004-05-19 06:14:06 +00:00
Stéphane Loeuillet
c31c96a7ea rename WMAx codec defines to be consistent
Original commit message from CVS:
rename WMAx codec defines to be consistent
2004-05-17 16:29:16 +00:00
Ronald S. Bultje
6e628ecf6d gst-libs/gst/audio/audioclock.c: Fix wrong return type (#142205).
Original commit message from CVS:
reviewed by: Ronald Bultje  <rbultje@ronald.bitfreak.net>
* gst-libs/gst/audio/audioclock.c:
Fix wrong return type (#142205).
2004-05-16 20:55:38 +00:00
Ronald S. Bultje
49c96eeb84 gst-libs/gst/riff/riff-read.c: Don't touch events after not owning them anymore.
Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_seek):
Don't touch events after not owning them anymore.
* gst/wavparse/gstwavparse.c: (gst_wavparse_base_init),
(gst_wavparse_fmt), (gst_wavparse_other),
(gst_wavparse_handle_seek), (gst_wavparse_loop),
(gst_wavparse_pad_convert), (gst_wavparse_pad_query),
(gst_wavparse_srcpad_event):
* gst/wavparse/gstwavparse.h:
Add seeking, fix querying.
2004-05-16 20:06:35 +00:00
Stéphane Loeuillet
7223edd775 gst-libs/gst/riff/riff-read.c: - fix INFO tag extraction in RIFF/AVI files because gst_event_unref (event) also freed...
Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c :
- fix INFO tag extraction in RIFF/AVI files
because gst_event_unref (event) also freed taglist
- avoid a mem leak
2004-05-14 17:23:08 +00:00
Stéphane Loeuillet
d1f0f46a3a - typos : unkown => unknown
Original commit message from CVS:
- typos : unkown => unknown
- missing break in RIFF/AVI iavs caps creation (was triggering a gsttag.c assert
2004-05-14 11:39:41 +00:00
Stéphane Loeuillet
dd2e09aa6b gst-libs/gst/riff/riff-ids.h: - add CDXA to the list of RIFF types
Original commit message from CVS:
* gst-libs/gst/riff/riff-ids.h :
- add CDXA to the list of RIFF types
- add plst (playlist ?) to wav chunk list (only diff with wavparse/riff.h)
2004-05-13 11:32:38 +00:00
Stéphane Loeuillet
2800919549 gst/auparse/gstauparse.c: eee (32, 64) only unsupported formats are ADPCM/CCITT G.72x
Original commit message from CVS:
* gst/auparse/gstauparse.c:
fixes a-law, adds mu-law, linear pcm (8,16,24,32), ieee (32, 64)
only unsupported formats are ADPCM/CCITT G.72x
reviewed by Ronald
* gst-libs/gst/audio/audio.h:
adds 24bit depth to PCM (x-raw-int)
2004-05-10 23:55:17 +00:00
Ronald S. Bultje
7180b58bb3 gst-libs/gst/riff/: Fix for unaligned RIFF files (i.e. where all the chunks together in a LIST chunk are not of the s...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data),
(gst_riff_create_audio_caps),
(gst_riff_create_audio_template_caps):
* gst-libs/gst/riff/riff-read.c: (gst_riff_peek_head):
Fix for unaligned RIFF files (i.e. where all the chunks together
in a LIST chunk are not of the same size as the size given in
the LIST chunk header). Fixes several odd WAVE files. Also fix
ADPCM (block_align property) in audio, so that wavparse based
on this works now as it used to stand-alone.
2004-05-09 15:49:25 +00:00
Iain Holmes
935969b8c1 Rewrote wavparse to use riff-read instead of doing bytestream stuff by hand.
Original commit message from CVS:
Rewrote wavparse to use riff-read instead of doing bytestream stuff by hand.
Made some useful functions in riff-read non-static.
2004-05-08 00:33:39 +00:00
Benjamin Otte
c17acf4260 gst-libs/gst/riff/riff-media.c: audio/x-raw-int with height rules! not. Now it's depth.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
audio/x-raw-int with height rules! not. Now it's depth.
2004-04-23 02:32:58 +00:00
Benjamin Otte
0f6862668b gst-libs/gst/riff/riff-media.c: mpegversion is an int
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data):
mpegversion is an int
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_base_init):
don't try to create pad templates with NULL caps, use any caps
instead.
2004-04-22 03:29:01 +00:00
David Schleef
df3fa511a7 configure.ac: bump required gstreamer version to 0.8.1.1 because of following changes [--ds]
Original commit message from CVS:
reviewed by David Schleef
* configure.ac: bump required gstreamer version to 0.8.1.1
because of following changes [--ds]
* gst-libs/gst/riff/riff-read.c:  Include gst/gstutils.h.
(gst_riff_peek_head, gst_riff_peek_list, gst_riff_read_list)
(gst_riff_read_header):  Use GST_READ_UINT*
macros to access possibly unaligned memory.
* gst/typefind/gsttypefindfunctions.c: Include gst/gstutils.h.
(mp3_type_find):  Use GST_READ_UINT*
macros to access possibly unaligned memory.
(mp3_type_find, mpeg1_parse_header, qt_type_find)
(speex_type_find): Likewise
* gst/tags/gstvorbistag.c: (ADVANCE): Likewise
* gst/qtdemux/qtdemux.c: Include stdlib.h (needed by realloc).
(QTDEMUX_GUINT32_GET, QTDEMUX_GUINT16_GET, QTDEMUX_FP32_GET)
(QTDEMUX_FP16_GET, QTDEMUX_FOURCC_GET)
(gst_qtdemux_loop_header, gst_qtdemux_loop_header)
(qtdemux_node_dump_foreach, qtdemux_tree_get_child_by_type)
(qtdemux_tree_get_sibling_by_type):  Use GST_READ_UINT*
macros to access possibly unaligned memory.
* gst/mpegstream/gstmpegpacketize.c: (parse_generic, parse_chunk):
Likewise.
* gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_parse_syshead)
(gst_mpeg_demux_parse_packet, gst_mpeg_demux_parse_pes): Likewise.
* gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_chain):
Likewise.
* gst/mpeg2sub/gstmpeg2subt.c: (GST_BUFFER_DATA)
(gst_mpeg2subt_chain_subtitle): Likewise.
* gst/mpeg1videoparse/gstmp1videoparse.c: (mp1videoparse_parse_seq)
(gst_mp1videoparse_time_code, gst_mp1videoparse_real_chain):
Likewise.
* gst/mpeg1sys/buffer.c: (mpeg1mux_buffer_update_audio_info):
Likewise.
* gst/cdxaparse/gstcdxaparse.c: (gst_bytestream_peek_bytes):
Likewise.
* gst/asfdemux/gstasfdemux.c: (_read_var_length, _read_uint):
Likewise.
2004-04-20 21:04:21 +00:00
David Schleef
3e84dfa539 gconf/Makefile.am: Fix for non-GNU make
Original commit message from CVS:
* gconf/Makefile.am: Fix for non-GNU make
* gst-libs/gst/Makefile.am: Change directory order to handle
GstPlay linking with gstinterfaces
* gst-libs/gst/audio/make_filter: make use of tr portable
* gst-libs/gst/play/Makefile.am: Add intended \
* gst-libs/gst/xwindowlistener/xwindowlistener.c:
(gst_xwin_set_clips): Switch to ISO variadic macro. Use a
function prototype instead of void *.
* gst/ffmpegcolorspace/gstffmpegcodecmap.c: Switch to ISO variadic
macro.
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcolorspace_chain): wrap NULL in GST_ELEMENT_ERROR call
* gst/videofilter/make_filter: make use of tr portable
* pkgconfig/Makefile.am: Remove GNU extension in Makefile target
2004-04-19 22:51:57 +00:00
Ronald S. Bultje
1c3add3521 gst-libs/gst/riff/riff-media.c: Add div[3456] as fourccs for DivX 3 (fixes #140137).
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data):
Add div[3456] as fourccs for DivX 3 (fixes #140137).
2004-04-16 01:43:13 +00:00
Ronald S. Bultje
38e6fccd41 gst/: Add MS RLE support. I added some functions to read out strf chunks into strf chunks and the data behind it. Thi...
Original commit message from CVS:
reviewed by: <delete if not using a buddy>
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data),
(gst_riff_create_video_caps), (gst_riff_create_audio_caps),
(gst_riff_create_video_template_caps),
(gst_riff_create_audio_template_caps):
* gst-libs/gst/riff/riff-media.h:
* gst-libs/gst/riff/riff-read.c:
(gst_riff_read_strf_vids_with_data), (gst_riff_read_strf_vids):
* gst-libs/gst/riff/riff-read.h:
* gst/avi/gstavidemux.c: (gst_avi_demux_add_stream):
Add MS RLE support. I added some functions to read out strf chunks
into strf chunks and the data behind it. This is usually color
palettes (as in RLE, but also in 8-bit RGB). Also use those during
caps creation. Lastly, add ADPCM (similar to wavparse - which
should eventually be rifflib based).
* gst/matroska/matroska-demux.c: (gst_matroska_demux_class_init),
(gst_matroska_demux_init), (gst_matroska_demux_reset):
* gst/matroska/matroska-demux.h:
Remove placeholders for some prehistoric tagging system. Didn't add
support for any tag system really anyway.
* gst/qtdemux/qtdemux.c:
Add support for audio/x-m4a (MPEG-4) through spider.
* gst/wavparse/gstwavparse.c: (gst_wavparse_parse_fmt),
(gst_wavparse_loop):
ADPCM support (#135862). Increase max. buffer size because we
cannot split buffers for ADPCM (screws references) and I've seen
files with 2048 byte chunks. 4096 seems safe for now.
2004-04-16 01:20:43 +00:00
Ronald S. Bultje
2d7d24378e gst-libs/gst/riff/riff-media.c: Fix typo in divxversion (3 instead of 4 for "DIVX" fourcc).
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
Fix typo in divxversion (3 instead of 4 for "DIVX" fourcc).
2004-04-15 03:21:14 +00:00
Ronald S. Bultje
7897fe29b9 gst-libs/gst/riff/riff-media.c: Add MS video v1.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
(gst_riff_create_video_template_caps):
Add MS video v1.
* gst/avi/gstavidemux.c: (gst_avi_demux_stream_index),
(gst_avi_demux_stream_data):
Add support for "rec-list" chunks.
2004-04-11 21:36:45 +00:00
Ronald S. Bultje
53d78b8a7b gst-libs/gst/riff/riff-media.c: Fix another codecname mismatch.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Fix another codecname mismatch.
2004-04-11 19:33:55 +00:00
Ronald S. Bultje
292a24d4cc gst-libs/gst/riff/riff-media.c: Fix divx caps mismatch and move from video/x-jpeg to image/jpeg so that MJPEG plays b...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
Fix divx caps mismatch and move from video/x-jpeg to image/jpeg
so that MJPEG plays back.
2004-04-11 17:21:32 +00:00
Benjamin Otte
253562933c gst-libs/gst/gconf/test-gconf.c: add missing gst_init
Original commit message from CVS:
* gst-libs/gst/gconf/test-gconf.c: (main):
add missing gst_init
2004-04-07 00:54:03 +00:00
Thomas Vander Stichele
195d892034 leak plugs and style fixes
Original commit message from CVS:
leak plugs and style fixes
2004-04-01 08:42:30 +00:00
Benjamin Otte
f9dbfb26cf try fixing configure on RH9
Original commit message from CVS:
try fixing configure on RH9
2004-03-30 03:13:03 +00:00
Thomas Vander Stichele
c65246f146 return proper values for state change failures
Original commit message from CVS:
return proper values for state change failures
2004-03-29 17:19:38 +00:00
Ronald S. Bultje
41182271ed ext/alsa/gstalsa.h: Remove unused entry.
Original commit message from CVS:
* ext/alsa/gstalsa.h:
Remove unused entry.
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
Add cinepak.
* gst/videodrop/gstvideodrop.c: (gst_videodrop_getcaps),
(gst_videodrop_link), (gst_videodrop_chain):
Fix, sort of. Was horribly broken with new capsnego. Bah...
2004-03-24 02:12:30 +00:00
Johan Dahlin
a4b140116c gst-libs/gst/play/play.c (gst_play_audio_fixate)
Original commit message from CVS:
* gst-libs/gst/play/play.c (gst_play_audio_fixate)
(gst_play_video_fixate): Check so the structure has the field
before trying to fixate them, this makes it possible to have
fakesinks for video and audio output without printing errors on
the output console.
2004-03-23 16:59:59 +00:00
David Schleef
3d743d8c3f gst-libs/gst/Makefile.am: Enable xoverlay unconditionally, since it doesn't depend on X, and it's part of our ABI.
Original commit message from CVS:
* gst-libs/gst/Makefile.am:  Enable xoverlay unconditionally,
since it doesn't depend on X, and it's part of our ABI.
2004-03-19 23:44:33 +00:00
Thomas Vander Stichele
8fa8a07c96 update libtool version actually use it in media-info
Original commit message from CVS:
update libtool version
actually use it in media-info
2004-03-15 23:31:46 +00:00
Julien Moutte
e71fc2718f gst-libs/gst/play/play.c: First draft of gst_play_get_framerate.
Original commit message from CVS:
2004-03-15  Julien MOUTTE <julien@moutte.net>

* gst-libs/gst/play/play.c: (gst_play_get_framerate),
(gst_play_get_sink_element): First draft of gst_play_get_framerate.
* gst-libs/gst/play/play.h:
2004-03-15 21:41:35 +00:00
Thomas Vander Stichele
f83cb187de don't mix tabs and spaces
Original commit message from CVS:
don't mix tabs and spaces
2004-03-15 19:32:28 +00:00
Thomas Vander Stichele
6a2a562f58 reverse order
Original commit message from CVS:
reverse order
2004-03-15 17:38:42 +00:00
Thomas Vander Stichele
269db252c7 some info
Original commit message from CVS:
some info
2004-03-15 17:19:06 +00:00
Thomas Vander Stichele
5dcc192fe5 adding media info pc files package them use ffmpegcolorspace in gstplay
Original commit message from CVS:
adding media info pc files
package them
use ffmpegcolorspace in gstplay
2004-03-15 17:17:28 +00:00
Johan Dahlin
bf07c7cc9b *.h: Revert indenting
Original commit message from CVS:
* *.h: Revert indenting
2004-03-15 16:32:55 +00:00
Thomas Vander Stichele
4df3f18839 gst-indent
Original commit message from CVS:
gst-indent
2004-03-14 22:34:34 +00:00
Benjamin Otte
51b07852e2 move gstreamer-gconf pkgconfig files to pkgconfig/ dir. Make sure they get rebuilt properly
Original commit message from CVS:
* configure.ac:
* gst-libs/gst/gconf/Makefile.am:
* pkgconfig/Makefile.am:
move gstreamer-gconf pkgconfig files to pkgconfig/ dir. Make sure
they get rebuilt properly
* configure.ac:
when checking for vorbis, try pkgconfig first.
* gst/modplug/gstmodplug.cc:
add fixate function
2004-03-14 10:33:44 +00:00
Benjamin Otte
19d221ba71 gst-libs/gst/xoverlay/xoverlay.*: replace XID with unsigned long to get rid of the xlibs dependency in
Original commit message from CVS:
* gst-libs/gst/xoverlay/xoverlay.c: (gst_x_overlay_set_xwindow_id),
(gst_x_overlay_got_xwindow_id):
* gst-libs/gst/xoverlay/xoverlay.h:
replace XID with unsigned long to get rid of the xlibs dependency in
XOverlay (fixes #137004)
2004-03-12 21:43:25 +00:00
Benjamin Otte
64157f4fc8 gst-libs/gst/mixer/mixer.c: the signals take 2 arguments
Original commit message from CVS:
* gst-libs/gst/mixer/mixer.c: (gst_mixer_class_init):
the signals take 2 arguments
2004-03-09 21:38:40 +00:00
David Schleef
826d9a9e66 ext/lcs/Makefile.am: Fix so that the lcs colorspace plugin doesn't conflict with the internal colorspace plugin.
Original commit message from CVS:
* ext/lcs/Makefile.am:  Fix so that the lcs colorspace plugin
doesn't conflict with the internal colorspace plugin.
* gst-libs/gst/audio/make_filter:  Use `` instead of $() to
satisfy the crappy-ass shell shipped by a certain vendor.
* gst/videofilter/make_filter: same (bug #135299)
2004-03-09 20:55:30 +00:00
Ronald S. Bultje
bd421d033a gst-libs/gst/riff/riff-read.c: Fix metadata read crash (#136537).
Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_info):
Fix metadata read crash (#136537).
2004-03-09 04:30:02 +00:00
Thomas Vander Stichele
5e57bf7de7 small media-info fixes
Original commit message from CVS:
small media-info fixes
2004-03-08 22:16:04 +00:00
Thomas Vander Stichele
0de1a7d73b compiler fixes
Original commit message from CVS:
compiler fixes
2004-03-07 21:05:40 +00:00
Thomas Vander Stichele
c0e6138393 compile fixes
Original commit message from CVS:
compile fixes
2004-03-07 21:02:25 +00:00
Thomas Vander Stichele
a9cf545d20 this should more or less work correctly.
Original commit message from CVS:
this should more or less work correctly.
2004-03-07 20:51:10 +00:00
Thomas Vander Stichele
8517f3b3c1 first pass at making this work again needs quite a lot of cleanups
Original commit message from CVS:
first pass at making this work again
needs quite a lot of cleanups
2004-03-06 20:15:17 +00:00
David Schleef
befdae8cda ext/faad/gstfaad.c: Fix negotiation.
Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_init), (gst_faad_srcgetcaps),
(gst_faad_chain): Fix negotiation.
* ext/librfb/gstrfbsrc.c: (gst_rfbsrc_handle_src_event): Add
key and button events.
* gst-libs/gst/floatcast/floatcast.h: Fix a minor bug in this
dung heap of code.
* gst-libs/gst/gconf/gstreamer-gconf-uninstalled.pc.in: gstgconf
depends on gconf
* gst-libs/gst/gconf/gstreamer-gconf.pc.in: same
* gst-libs/gst/play/play.c: (gst_play_pipeline_setup),
(gst_play_video_fixate), (gst_play_audio_fixate): Add a fixate
function to encourage better negotiation, particularly between
audioconvert and osssink.
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain):
* gst/qtdemux/qtdemux.c: (qtdemux_parse_trak):  Make some debugging
more important.
* gst/typefind/gsttypefindfunctions.c:  Fix mistake in flash
typefinding.
* gst/vbidec/vbiscreen.c:  Add glib header
* pkgconfig/gstreamer-play.pc.in:  Depends on gst-interfaces.
2004-03-06 00:42:20 +00:00
Thomas Vander Stichele
bc85c3da12 fix gconf install path
Original commit message from CVS:
fix gconf install path
2004-03-05 12:13:10 +00:00
Thomas Vander Stichele
15f1314023 fix headers fix signals to use - instead of _
Original commit message from CVS:
fix headers
fix signals to use - instead of _
2004-03-04 21:39:15 +00:00
Julien Moutte
b08569a5fc gst-libs/gst/gconf/gconf.c: Using ximagesink as a default if no gconf key found. We should probably consider using al...
Original commit message from CVS:
2004-03-03  Julien MOUTTE <julien@moutte.net>

* gst-libs/gst/gconf/gconf.c: (gst_gconf_get_default_video_sink):
Using ximagesink as a default if no gconf key found. We should
probably consider using alsasink instead of osssink for the audio
part.
2004-03-03 11:41:50 +00:00
Thomas Vander Stichele
efaac02d08 xwindowlistener is a plugin helper lib, not a full lib
Original commit message from CVS:
xwindowlistener is a plugin helper lib, not a full lib
2004-03-01 23:20:52 +00:00
Thomas Vander Stichele
2279068b51 fix plugin names move gstvideosink.h to videosink.h
Original commit message from CVS:
fix plugin names
move gstvideosink.h to videosink.h
2004-03-01 16:17:21 +00:00
Benjamin Otte
e4a3aa6fba gst-libs/gst/audio/audio.h: add macro to make sure header isn't included twice
Original commit message from CVS:
2004-02-27  Benjamin Otte  <otte@gnome.org>

* gst-libs/gst/audio/audio.h:
add macro to make sure header isn't included twice
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_process_chunk):
don't use gst_buffer_free
* gst/playondemand/filter.func:
don't usae gst_data_free. Free data only once.
2004-02-27 21:47:49 +00:00
David Schleef
14db016878 gst/: -marshal.[ch] and -enum.[ch] files should not be disted, -marshal.h files should not be installed, and -enum.h ...
Original commit message from CVS:
* gst-libs/gst/colorbalance/Makefile.am:
* gst-libs/gst/mixer/Makefile.am:
* gst-libs/gst/tuner/Makefile.am:
* gst/level/Makefile.am: -marshal.[ch] and -enum.[ch] files
should not be disted, -marshal.h files should not be installed,
and -enum.h files _should_ be installed.  Fix to make this the
case.
2004-02-27 02:39:18 +00:00
Johan Dahlin
cc02b0dd3c gst-libs/gst/gconf/gconf.c: Add \n to g_print error messages
Original commit message from CVS:
* gst-libs/gst/gconf/gconf.c: Add \n to g_print error messages
2004-02-24 17:16:07 +00:00
Thomas Vander Stichele
ff81abd2bf versioning gstreamer schemas and install location
Original commit message from CVS:
versioning gstreamer schemas and install location
2004-02-23 22:31:48 +00:00
Benjamin Otte
409ddc4500 ext/xine/xineinput.c: call parent dispose.
Original commit message from CVS:
2004-02-23  Benjamin Otte  <otte@gnome.org>

* ext/xine/xineinput.c: (gst_xine_input_dispose):
(gst_xine_input_subclass_init):
call parent dispose.
change pad template for CD reader correctly
* ext/xine/Makefile.am:
* ext/xine/gstxine.h:
* ext/xine/xine.c: (plugin_init):
* ext/xine/xineaudiosink.c:
wrap audio sinks, too
* gst-libs/gst/resample/private.h:
* gst-libs/gst/resample/resample.c: (gst_resample_init),
(gst_resample_reinit), (gst_resample_scale),
(gst_resample_nearest_s16), (gst_resample_bilinear_s16),
(gst_resample_sinc_slow_s16), (gst_resample_sinc_s16),
(gst_resample_sinc_ft_s16), (gst_resample_nearest_float),
(gst_resample_bilinear_float), (gst_resample_sinc_slow_float),
(gst_resample_sinc_float), (gst_resample_sinc_ft_float):
* gst-libs/gst/resample/resample.h:
* gst/audioscale/gstaudioscale.c: (gst_audioscale_method_get_type),
(gst_audioscale_class_init), (gst_audioscale_link),
(gst_audioscale_get_buffer), (gst_audioscale_init),
(gst_audioscale_chain), (gst_audioscale_set_property),
(gst_audioscale_get_property):
* gst/audioscale/gstaudioscale.h:
s/resample_*/gst_resample_*/i to not clobber namespaces
2004-02-23 22:21:30 +00:00
Julien Moutte
678a8ee805 gst/: First batch implementing audio and video codec tags in demuxers.
Original commit message from CVS:
2004-02-23  Julien MOUTTE  <julien@moutte.net>

* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
(gst_riff_create_audio_caps), (gst_riff_create_iavs_caps),
(gst_riff_create_video_template_caps),
(gst_riff_create_audio_template_caps),
(gst_riff_create_iavs_template_caps):
* gst-libs/gst/riff/riff-media.h:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_base_init),
(gst_asf_demux_audio_caps), (gst_asf_demux_add_audio_stream),
(gst_asf_demux_video_caps), (gst_asf_demux_add_video_stream):
* gst/avi/gstavidemux.c: (gst_avi_demux_add_stream):
* gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream),
(gst_matroska_demux_video_caps), (gst_matroska_demux_audio_caps),
(gst_matroska_demux_plugin_init): First batch implementing audio and
video codec tags in demuxers.
2004-02-23 22:16:21 +00:00
Thomas Vander Stichele
38471bcd13 break up link function for better error debug
Original commit message from CVS:
break up link function for better error debug
2004-02-18 18:41:36 +00:00
Julien Moutte
e678129a7e examples/gstplay/player.c: Adding some output for debugging.
Original commit message from CVS:
2004-02-15  Julien MOUTTE  <julien@moutte.net>

* examples/gstplay/player.c: (got_eos), (main): Adding some
output for debugging.
* gst-libs/gst/play/play.c: (gst_play_state_change): Stop our
timeouts if we go to any state different from PLAYING.
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_seek): Fix some
more EOS bugs in riff lib.
2004-02-15 17:41:28 +00:00
Julien Moutte
a21475966d gst-libs/gst/play/play.c: Disable visualization until i find a way to fix switch correctly.
Original commit message from CVS:
2004-02-14  Julien MOUTTE  <julien@moutte.net>

* gst-libs/gst/play/play.c: (gst_play_connect_visualization): Disable
visualization until i find a way to fix switch correctly.
* gst-libs/gst/riff/riff-read.c: (gst_riff_peek_head): Fix a bug when
EOS arrives.
* gst/switch/gstswitch.c: (gst_switch_release_pad),
(gst_switch_request_new_pad), (gst_switch_poll_sinkpads),
(gst_switch_loop), (gst_switch_dispose), (gst_switch_class_init):
Reworked switch to get a more correct behaviour with events and refing
of data stored in sinkpads.
* gst/switch/gstswitch.h: Adding an eos flag for every sinkpad so that
we don't pull from a pad in EOS.
2004-02-14 18:37:48 +00:00
Thomas Vander Stichele
b0ecd5f659 move ffmpeg stuff to gst-ffmpeg module
Original commit message from CVS:
move ffmpeg stuff to gst-ffmpeg module
2004-02-13 15:11:50 +00:00
Benjamin Otte
07674249eb gst-libs/gst/: remove unused GST_OPT_CFLAGS from Makefiles include X_CFLAGS and X_LIBS in xoverlay. (#131948)
Original commit message from CVS:
2004-02-08  Benjamin Otte  <in7y118@public.uni-hamburg.de>

* gst-libs/gst/colorbalance/Makefile.am:
* gst-libs/gst/navigation/Makefile.am:
* gst-libs/gst/xoverlay/Makefile.am:
remove unused GST_OPT_CFLAGS from Makefiles
include X_CFLAGS and X_LIBS in xoverlay. (#131948)
2004-02-08 13:35:42 +00:00
David Schleef
f69822bcbd Build fixes for OS X: (see #129600)
Original commit message from CVS:
Build fixes for OS X: (see #129600)
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_strh),
(gst_riff_read_strf_vids), (gst_riff_read_strf_auds),
(gst_riff_read_strf_iavs):
* gst/avi/gstavidemux.c: (gst_avi_demux_stream_avih),
(gst_avi_demux_stream_odml):
* gst/playondemand/Makefile.am:
* gst/rtp/rtp-packet.c:
2004-02-07 01:34:04 +00:00
Benjamin Otte
9b47d6474a gst-libs/gst/riff/riff-read.c: use correct GST_TAG_ENCODER tag
Original commit message from CVS:
2004-02-05  Benjamin Otte  <in7y118@public.uni-hamburg.de>

* gst-libs/gst/riff/riff-read.c: (gst_riff_read_info):
use correct GST_TAG_ENCODER tag
2004-02-05 03:02:42 +00:00
Julien Moutte
d1f51a2773 gst-libs/gst/xoverlay/xoverlay.*: Adding the _expose method to tell an overlay to redraw the image because it has bee...
Original commit message from CVS:
2004-02-04  Julien MOUTTE  <julien@moutte.net>

* gst-libs/gst/xoverlay/xoverlay.c: (gst_x_overlay_expose):
* gst-libs/gst/xoverlay/xoverlay.h: Adding the _expose method to tell
an overlay to redraw the image because it has been exposed.
* sys/ximage/ximagesink.c: (gst_ximagesink_ximage_destroy),
(gst_ximagesink_ximage_put), (gst_ximagesink_expose),
(gst_ximagesink_xoverlay_init), (gst_ximagesink_init):
* sys/ximage/ximagesink.h: Implement expose method from XOverlay
interface
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_destroy),
(gst_xvimagesink_xvimage_put), (gst_xvimagesink_expose),
(gst_xvimagesink_xoverlay_init), (gst_xvimagesink_init):
* sys/xvimage/xvimagesink.h: Implement expose method from XOverlay
interface
2004-02-03 23:05:46 +00:00
Thomas Vander Stichele
aa67552268 last glib_root standing
Original commit message from CVS:
last glib_root standing
2004-02-03 20:27:24 +00:00
Thomas Vander Stichele
a9ec91658e remove glib_root add hack rule for .Plo files
Original commit message from CVS:
remove glib_root
add hack rule for .Plo files
2004-02-03 20:20:07 +00:00
Thomas Vander Stichele
e963f78788 use glib-gen.mak fix Makefile.am bugs and clean them up
Original commit message from CVS:
use glib-gen.mak
fix Makefile.am bugs and clean them up
2004-02-03 16:49:54 +00:00
David I. Lehn
95778bdb9f gst-libs/gst/: Generate enum type code with glib-mkenums.
Original commit message from CVS:
* gst-libs/gst/colorbalance/Makefile.am:
* gst-libs/gst/colorbalance/colorbalance.h:
* gst-libs/gst/mixer/Makefile.am:
* gst-libs/gst/mixer/mixer.h:
* gst-libs/gst/play/Makefile.am:
* gst-libs/gst/play/play.h:
* gst-libs/gst/tuner/Makefile.am:
* gst-libs/gst/tuner/tuner.h:
Generate enum type code with glib-mkenums.
* gst-libs/gst/colorbalance/.cvsignore:
* gst-libs/gst/mixer/.cvsignore:
* gst-libs/gst/play/.cvsignore:
* gst-libs/gst/tuner/.cvsignore:
Ignore generated files.
2004-02-03 08:56:06 +00:00
David I. Lehn
84e5239830 gst-libs/gst/audio/.cvsignore: Ignore generated file.
Original commit message from CVS:
* gst-libs/gst/audio/.cvsignore:
Ignore generated file.
* gst-libs/gst/audio/Makefile.am:
Do not install example filter.
2004-02-03 08:52:58 +00:00
Julien Moutte
2849678421 gst-libs/gst/play/play.c: Moving volume in the audio thread for instantaneous volume change. Maybe i will add another...
Original commit message from CVS:
2004-02-02  Julien MOUTTE  <julien@moutte.net>

* gst-libs/gst/play/play.c: (gst_play_pipeline_setup),
(gst_play_set_audio_sink): Moving volume in the audio thread for
instantaneous volume change. Maybe i will add another volume in front
of visualization later, not sure yet though.
2004-02-02 22:18:38 +00:00
David Schleef
e4be2582bc code cleanup. Change bzero() to memset(). Remove duplicate ; at ends of lines.
Original commit message from CVS:
code cleanup.  Change bzero() to memset().  Remove duplicate ; at ends
of lines.
* ext/cdparanoia/gstcdparanoia.c: (cdparanoia_event):
* ext/flac/gstflactag.c: (gst_flac_tag_chain):
* ext/xvid/gstxviddec.c: (gst_xviddec_src_link):
* gst-libs/gst/play/play.c: (gst_play_get_sink_element):
* gst/ac3parse/gstac3parse.c: (gst_ac3parse_chain):
* gst/effectv/gstedge.c: (gst_edgetv_sinkconnect):
* gst/effectv/gstvertigo.c: (gst_vertigotv_sinkconnect):
* gst/intfloat/float22int.c: (gst_float2_2_int_getcaps),
(gst_float2_2_int_link):
* gst/mpeg2sub/gstmpeg2subt.c: (gst_mpeg2subt_chain_subtitle):
* gst/rtjpeg/RTjpeg.c: (RTjpeg_init_mcompress):
* gst/tcp/gsttcpsink.c: (gst_tcpsink_init_send):
* gst/tcp/gsttcpsrc.c: (gst_tcpsrc_init_receive):
* gst/udp/gstudpsink.c: (gst_udpsink_init_send):
* gst/udp/gstudpsrc.c: (gst_udpsrc_init_receive):
* sys/v4l/gstv4lelement.c: (gst_v4lelement_init):
* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_set_capture):
* testsuite/gst-lint: Add tests for bzero and ;;
2004-02-02 20:09:09 +00:00
Thomas Vander Stichele
8336c47f0c change NULL to (NULL) for GST_ELEMENT_ERROR
Original commit message from CVS:
change NULL to (NULL) for GST_ELEMENT_ERROR
Make sure errors end with "."
2004-02-02 17:23:33 +00:00
Thomas Vander Stichele
ec00f91abb block tick callback for 0.5 sec after a seek
Original commit message from CVS:
block tick callback for 0.5 sec after a seek
2004-02-02 10:49:32 +00:00
Thomas Vander Stichele
1474d866bc check for NULLness and warn if we can't report GError
Original commit message from CVS:
check for NULLness and warn if we can't report GError
2004-02-02 10:19:38 +00:00
Julien Moutte
6300c130df gst-libs/gst/play/play.c: s/sinkconnect/sink_link
Original commit message from CVS:
2004-02-01  Julien MOUTTE  <julien@moutte.net>

* gst-libs/gst/play/play.c: (gst_play_seek_to_time),
(gst_play_new): Accepting NULL GError, blocking time tick while seeking.        * sys/ximage/ximagesink.c: (gst_ximagesink_sink_link),
(gst_ximagesink_chain), (gst_ximagesink_init): s/sinkconnect/sink_link
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_sink_link),
(gst_xvimagesink_chain), (gst_xvimagesink_init): s/sinkconnect/sink_link
2004-02-01 22:33:07 +00:00
Thomas Vander Stichele
e2469ac793 add error handling
Original commit message from CVS:
add error handling
2004-01-31 15:56:32 +00:00
Thomas Vander Stichele
b106d090bc done operating
Original commit message from CVS:
done operating
2004-01-30 22:38:56 +00:00
Thomas Vander Stichele
6dd220eaac removing deprecated code
Original commit message from CVS:
removing deprecated code
2004-01-30 22:03:14 +00:00
Ronald S. Bultje
3e1a27491e gst/: Fix event handling.
Original commit message from CVS:
2004-01-30  Ronald Bultje  <rbultje@ronald.bitfreak.net>

* gst-libs/gst/riff/riff-read.c: (gst_riff_peek_head),
(gst_riff_read_seek):
* gst/matroska/ebml-read.c: (gst_ebml_read_element_id),
(gst_ebml_read_seek):
Fix event handling.
2004-01-30 12:37:58 +00:00
Thomas Vander Stichele
03caa97b73 GST_ELEMENT_ERROR
Original commit message from CVS:
GST_ELEMENT_ERROR
2004-01-29 23:20:45 +00:00
Julien Moutte
1e9f5c82be ext/gnomevfs/gstgnomevfssrc.c: Fixing seeking emiting FLUSH and even before DISCONT.
Original commit message from CVS:
2004-01-29  Julien MOUTTE  <julien@moutte.net>

* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_get): Fixing seeking
emiting FLUSH and even before DISCONT.
* gst-libs/gst/play/gstplay.c: (gst_play_seek_to_time): Fix seeking to
get the best instant seeking as possible yay!
2004-01-29 22:44:42 +00:00
Julien Moutte
f606249297 gst-libs/gst/play/gstplay.c: seek on video sink element first.
Original commit message from CVS:
2004-01-28  Julien MOUTTE  <julien@moutte.net>

* gst-libs/gst/play/gstplay.c: (gst_play_seek_to_time): seek on video
sink element first.
* gst/videoscale/gstvideoscale.c:
(gst_videoscale_handle_src_event): Fixing src event handler.
2004-01-28 22:14:14 +00:00
Julien Moutte
b2fcdfe132 gst-libs/gst/play/gstplay.c: Fixing the way to get current position.
Original commit message from CVS:
2004-01-27  Julien MOUTTE <julien@moutte.net>

* gst-libs/gst/play/gstplay.c: (gst_play_tick_callback),
(gst_play_seek_to_time): Fixing the way to get current position.
2004-01-27 21:24:14 +00:00
Julien Moutte
ce1f3bd133 gst-libs/gst/play/gstplay.c: The easiest fix ever... Inverting 2 lines of code make spider autoplug correctly tagged ...
Original commit message from CVS:
2004-01-27  Julien MOUTTE <julien@moutte.net>

* gst-libs/gst/play/gstplay.c: (gst_play_set_location): The easiest
fix ever... Inverting 2 lines of code make spider autoplug correctly
tagged mp3 !
2004-01-27 16:42:40 +00:00
Benjamin Otte
b552a2d9f0 gst-libs/gst/audio/audio.h: remove buffer-frames from audio caps
Original commit message from CVS:
2004-01-26  Benjamin Otte  <in7y118@public.uni-hamburg.de>

* gst-libs/gst/audio/audio.h:
remove buffer-frames from audio caps
* gst/audioconvert/gstaudioconvert.c:
fix plugin to really work.
2004-01-26 03:54:21 +00:00
Ronald S. Bultje
7c1c3a173f gst-libs/gst/: Add gtk-doc style comments. Also fix a function name.
Original commit message from CVS:
2004-01-25  Ronald Bultje  <rbultje@ronald.bitfreak.net>

* gst-libs/gst/mixer/mixer.c:
* gst-libs/gst/propertyprobe/propertyprobe.c:
* gst-libs/gst/tuner/tuner.c: (gst_tuner_find_norm_by_name),
(gst_tuner_find_channel_by_name):
* gst-libs/gst/tuner/tuner.h:
Add gtk-doc style comments. Also fix a function name.
2004-01-25 22:07:16 +00:00
Julien Moutte
ff322c28e2 gst-libs/gst/play/gstplay.c: Another try in visualization implementation. Still have an issue with switch blocking wh...
Original commit message from CVS:
2004-01-25  Julien MOUTTE  <julien@moutte.net>

* gst-libs/gst/play/gstplay.c: (gst_play_pipeline_setup),
(gst_play_identity_handoff), (gst_play_set_location),
(gst_play_set_visualization), (gst_play_connect_visualization): Another
try in visualization implementation. Still have an issue with switch
blocking when pulling from video_queue and only audio comes out of
spider.
* gst/switch/gstswitch.c: (gst_switch_release_pad),
(gst_switch_poll_sinkpads), (gst_switch_class_init): Implementing pad
release method. And check if the pad is usable before pulling.
2004-01-25 12:28:05 +00:00
Ronald S. Bultje
20af0be15e gst-libs/gst/riff/riff-read.c: Additional pad usability check.
Original commit message from CVS:
2004-01-25  Ronald Bultje  <rbultje@ronald.bitfreak.net>

* gst-libs/gst/riff/riff-read.c: (gst_riff_read_info):
Additional pad usability check.
* gst/mpeg1videoparse/gstmp1videoparse.c: (gst_mp1videoparse_init),
(mp1videoparse_find_next_gop), (gst_mp1videoparse_time_code),
(gst_mp1videoparse_real_chain):
Fix MPEG video stream parsing. The original plugin had several
issues, including not timestamping streams where the source was
not timestamped (this happens with PTS values in mpeg system
streams, but MPEG video is also a valid stream on its own so
that needs timestamps too). We use the display time code for that
for now. Also, if one incoming buffer contains multiple valid
frames, we push them all on correctly now, including proper EOS
handling. Lastly, several potential segfaults were fixed, and we
properly sync on new sequence/gop headers to include them in next,
not previous frames (since they're header for the next frame, not
the previous). Also see #119206.
* gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_chain),
(bpf_from_header):
Move caps setting so we only do it after finding several valid
MPEG-1 fraes sequentially, not right after the first one (which
might be coincidental).
* gst/typefind/gsttypefindfunctions.c: (mpeg1_sys_type_find),
(mpeg_video_type_find), (mpeg_video_stream_type_find),
(plugin_init):
Add unsynced MPEG video stream typefinding, and change some
probability values so we detect streams rightly. The idea is as
follows: I can have an unsynced system stream which contains
video. In the current code, I would randomly get a type for either
system or video stream type found, because the probabilities are
being calculated rather randomly. I now use fixed values, so we
always prefer system stream if that was found (and that is how it
should be). If no system stream was found, we can still identity
the stream as video-only.
2004-01-25 00:25:16 +00:00
Ronald S. Bultje
163bceae95 gst-libs/gst/riff/riff-read.*: Remove stuff fromold metadata system.
Original commit message from CVS:
2004-01-23  Ronald Bultje  <rbultje@ronald.bitfreak.net>

* gst-libs/gst/riff/riff-read.c: (gst_riff_read_class_init),
(gst_riff_read_init), (gst_riff_read_change_state):
* gst-libs/gst/riff/riff-read.h:
Remove stuff fromold metadata system.
2004-01-23 14:03:16 +00:00
Ronald S. Bultje
fec321f575 ext/ogg/gstoggdemux.c: Fix wrong file comment.
Original commit message from CVS:
2004-01-23  Ronald Bultje  <rbultje@ronald.bitfreak.net>

* ext/ogg/gstoggdemux.c:
Fix wrong file comment.
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_info):
* gst/avi/gstavidemux.c: (gst_avi_demux_stream_header):
Add metadata reading properly.
2004-01-23 13:22:17 +00:00
Julien Moutte
15289fa2dd gst-libs/gst/play/gstplay.c: Reworked the pipeline from scratch. Visualization is back and switch went out as i reali...
Original commit message from CVS:
2004-01-23  Julien MOUTTE  <julien@moutte.net>

* gst-libs/gst/play/gstplay.c: (gst_play_pipeline_setup),
(gst_play_set_location), (gst_play_seek_to_time),
(gst_play_set_audio_sink), (gst_play_set_visualization),
(gst_play_connect_visualization), (gst_play_get_sink_element): Reworked
the pipeline from scratch. Visualization is back and switch went out as
i realized it was not possible to use the way i wanted.
* sys/ximage/ximagesink.c: (gst_ximagesink_imagepool_clear),
(gst_ximagesink_change_state), (gst_ximagesink_dispose): Move xcontext
clearing in state change from READY to NULL. So that one can clean the
X ressources keeping the element.
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_get),
(gst_xvimagesink_imagepool_clear), (gst_xvimagesink_change_state),
(gst_xvimagesink_colorbalance_set_value),
(gst_xvimagesink_colorbalance_get_value),
(gst_xvimagesink_set_property), (gst_xvimagesink_dispose),
(gst_xvimagesink_init): Same xcontext cleaning than ximagesink in state
change from READY to NULL and fixed some stupid bugs in colorbalance
get/set values. Also added the following feature : when nobody tries to
set some values to the colorbalance levels before the xcontext is
grabbed, then when creating channels list from Xv attributes we set the
internal values to the Xv defaults. This way we handle buggy Xv drivers
that set default hue values far from the middle of the range (Thanks
to Jon Trowbridge for pointing that issue).
* sys/xvimage/xvimagesink.h: Adding a cb_changed boolean to know if
colorbalance levels have been set before xcontext is grabbed.
2004-01-22 23:54:34 +00:00
Thomas Vander Stichele
b469416f53 fix up media-info now reports format again metadata needs some rewriting
Original commit message from CVS:
fix up media-info
now reports format again
metadata needs some rewriting
2004-01-22 19:00:27 +00:00
Thomas Vander Stichele
fd060aa027 register and use debugging category in media-info make vorbisfile emit tags for streaminfo
Original commit message from CVS:
register and use debugging category in media-info
make vorbisfile emit tags for streaminfo
2004-01-20 16:25:11 +00:00
Thomas Vander Stichele
9698ad445b fix distcheck
Original commit message from CVS:
fix distcheck
2004-01-20 11:25:09 +00:00
Thomas Vander Stichele
eefcaf5b9e adding locale.h
Original commit message from CVS:
adding locale.h
2004-01-19 16:43:01 +00:00
Thomas Vander Stichele
01e210d650 removing config.h
Original commit message from CVS:
removing config.h
2004-01-19 15:50:38 +00:00
Thomas Vander Stichele
0d030dd3fa putting i18n in place for plugins
Original commit message from CVS:
putting i18n in place for plugins
2004-01-19 15:45:55 +00:00
Thomas Vander Stichele
05abb4879e adding header
Original commit message from CVS:
adding header
2004-01-18 23:21:55 +00:00
Thomas Vander Stichele
b95a710d41 use new error signal and classification
Original commit message from CVS:
use new error signal and classification
2004-01-18 21:46:58 +00:00
Ronald S. Bultje
285797683c gst-libs/gst/Makefile.am: restructure so having local patches works easier.
Original commit message from CVS:
2004-01-16  Ronald Bultje  <rbultje@ronald.bitfreak.net>

* gst-libs/gst/Makefile.am:
restructure so having local patches works easier.
2004-01-16 14:55:44 +00:00
David Schleef
5dca8edfe9 gst-libs/gst/audio/Makefile.am: Add gstaudiofiltertemplate.c and building of gstaudiofilterexample.c from the template.
Original commit message from CVS:
* gst-libs/gst/audio/Makefile.am:
Add gstaudiofiltertemplate.c and building of gstaudiofilterexample.c
from the template.
* gst-libs/gst/audio/gstaudiofilter.c:
* gst-libs/gst/audio/gstaudiofilter.h:
Add bytes_per_sample and size and n_samples calculation.
* gst-libs/gst/audio/gstaudiofilterexample.c:
Remove, now autogenerated.
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
Moved from gstaudiofilterexample, object name changed, code added
so that it actually works.
* gst-libs/gst/audio/make_filter:
Script to build an audiofilter subclass from the template.
* gst/colorspace/Makefile.am:
* gst/colorspace/yuv2yuv.c:
Remove file, since it's GPL, and we don't use it.
2004-01-16 00:55:29 +00:00
Julien Moutte
9aa32817d9 ext/alsa/gstalsamixer.c: Setting mixer interface type to HARDWARE.
Original commit message from CVS:
2004-01-15  Julien MOUTTE <julien@moutte.net>

* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_interface_init): Setting
mixer interface type to HARDWARE.
* gst-libs/gst/mixer/mixer.c: (gst_mixer_class_init): Adding a default
type to SOFTWARE.
* gst-libs/gst/mixer/mixer.h: Adding mixer interface type and macro.
* gst-libs/gst/mixer/mixertrack.h: Adding mixertrack flag SOFTWARE.
* gst/volume/gstvolume.c: (gst_volume_interface_supported),
(gst_volume_interface_init), (gst_volume_list_tracks),
(gst_volume_set_volume), (gst_volume_get_volume),
(gst_volume_set_mute), (gst_volume_mixer_init),
(gst_volume_dispose), (gst_volume_get_type), (volume_class_init),
(volume_init): Implementing mixer interface.
* gst/volume/gstvolume.h: Adding tracklist for mixer interface.
* sys/oss/gstosselement.c: (gst_osselement_get_type),
(gst_osselement_change_state): Removing some trailing commas in
structures.
* sys/oss/gstossmixer.c: (gst_ossmixer_interface_init): Setting mixer
interface type to HARDWARE.
* sys/v4l/gstv4lcolorbalance.c:
(gst_v4l_color_balance_interface_init): Setting colorbalance interface
type to HARDWARE.
* sys/v4l2/gstv4l2colorbalance.c:
(gst_v4l2_color_balance_interface_init): Setting colorbalance
interface type to HARDWARE.
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_chain): use exactly the
same code than ximagesink for event handling.
2004-01-15 15:50:10 +00:00
Julien Moutte
ab07e8f25a gst-libs/gst/play/gstplay.c: Preparing switch integration, adding videobalance in the pipeline.
Original commit message from CVS:
2004-01-14  Julien MOUTTE  <julien@moutte.net>

* gst-libs/gst/play/gstplay.c: (gst_play_pipeline_setup),
(gst_play_set_location), (gst_play_set_visualization): Preparing
switch integration, adding videobalance in the pipeline.
2004-01-14 21:50:39 +00:00
Julien Moutte
2e376792cb gst-libs/gst/colorbalance/colorbalance.c: Adding a default type.
Original commit message from CVS:
2004-01-14  Julien MOUTTE <julien@moutte.net>

* gst-libs/gst/colorbalance/colorbalance.c:
(gst_color_balance_class_init): Adding a default type.
* gst-libs/gst/colorbalance/colorbalance.h: Adding a macro to access
the type.
* gst/videofilter/gstvideobalance.c: (gst_videobalance_get_type),
(gst_videobalance_dispose), (gst_videobalance_class_init),
(gst_videobalance_init), (gst_videobalance_interface_supported),
(gst_videobalance_interface_init),
(gst_videobalance_colorbalance_list_channels),
(gst_videobalance_colorbalance_set_value),
(gst_videobalance_colorbalance_get_value),
(gst_videobalance_colorbalance_init): Implementing colorbalance
interface.
* gst/videofilter/gstvideobalance.h: Adding colorbalance channels
list.
* sys/ximage/ximagesink.c: (gst_ximagesink_set_xwindow_id): Fixing a
bug which was triggering a BadAccess X error when setting an overlay
before pad was really negotiated.
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_colorbalance_init):
Using the colorbalance type macro.
2004-01-14 17:15:34 +00:00
Benjamin Otte
11c4fea0c8 gst-libs/gst/play/gstplay.c: Query the audio element to get the time, not the clock. We're interested in the element'...
Original commit message from CVS:
2004-01-14  Benjamin Otte  <in7y118@public.uni-hamburg.de>

* gst-libs/gst/play/gstplay.c: (gst_play_tick_callback):
Query the audio element to get the time, not the clock. We're
interested in the element's time here.
2004-01-14 02:37:12 +00:00
Benjamin Otte
3a64b7c99f use element time.
Original commit message from CVS:
2004-01-14  Benjamin Otte  <in7y118@public.uni-hamburg.de>

* ext/aalib/gstaasink.c: (gst_aasink_chain):
* ext/alsa/gstalsasink.c: (gst_alsa_sink_check_event):
* ext/esd/esdsink.c: (gst_esdsink_chain):
* ext/libcaca/gstcacasink.c: (gst_cacasink_chain):
* ext/mas/massink.c: (gst_massink_chain):
* ext/sdl/sdlvideosink.c: (gst_sdlvideosink_chain):
* gst/matroska/matroska-demux.c: (gst_matroska_demux_parse_index),
(gst_matroska_demux_parse_metadata):
* gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_loop),
(gst_mpeg_parse_release_locks):
* gst/tcp/gsttcpsink.c: (gst_tcpsink_chain):
* gst/udp/gstudpsink.c: (gst_udpsink_chain):
* gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_get):
* sys/oss/gstosssink.c: (gst_osssink_init), (gst_osssink_chain),
(gst_osssink_change_state):
* sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_chain):
* sys/ximage/ximagesink.c: (gst_ximagesink_chain):
* sys/xvideo/xvideosink.c: (gst_xvideosink_chain),
(gst_xvideosink_release_locks):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_chain):
use element time.
* ext/alsa/gstalsaclock.c: (gst_alsa_clock_start),
(gst_alsa_clock_stop):
* gst-libs/gst/audio/audioclock.c: (gst_audio_clock_set_active),
(gst_audio_clock_get_internal_time):
simplify for use with new clocking code.
* testsuite/alsa/Makefile.am:
* testsuite/alsa/sinesrc.c: (sinesrc_init), (sinesrc_force_caps):
fix testsuite for new caps system
2004-01-14 00:53:52 +00:00
Jon Trowbridge
0aef65d54e gst-libs/gst/colorbalance/colorbalance.h: Adding a type to the colorbalance interface stating if it is hardware based...
Original commit message from CVS:
2004-01-13  Julien MOUTTE <julien@moutte.net>

* gst-libs/gst/colorbalance/colorbalance.h: Adding a type to the
colorbalance interface stating if it is hardware based or software
based.
* gst/videofilter/gstvideobalance.c: (gst_videobalance_planar411):
Removing a trailing comma.
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_get),
(gst_xvimagesink_colorbalance_init): Integrating a patch from Jon
Trowbridge <trow@ximian.com> querying Xv adaptor for min/max value as
the documentation seems to be wrong on the -1000 to 1000 interval.
2004-01-13 17:57:25 +00:00
Julien Moutte
593632083c gst-libs/gst/colorbalance/colorbalancechannel.c: Adding safety check in dispose method.
Original commit message from CVS:
2004-01-12  Julien MOUTTE <julien@moutte.net>

* gst-libs/gst/colorbalance/colorbalancechannel.c:
(gst_color_balance_channel_dispose): Adding safety check in dispose
method.
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_get),
(gst_xvimagesink_xcontext_clear),
(gst_xvimagesink_interface_supported),
(gst_xvimagesink_colorbalance_list_channels),
(gst_xvimagesink_colorbalance_set_value),
(gst_xvimagesink_colorbalance_get_value),
(gst_xvimagesink_colorbalance_init), (gst_xvimagesink_get_type):
Adding colorbalance interface support to set XV parameters such as
HUE, BRIGHTNESS, CONTRAST, SATURATION.
* sys/xvimage/xvimagesink.h: Adding the channels list for colorbalance
interface.
2004-01-12 19:50:29 +00:00
Thomas Vander Stichele
5c63b06853 adding structure setters matching the templates for audio
Original commit message from CVS:
adding structure setters matching the templates for audio
2004-01-12 19:35:54 +00:00
Julien Moutte
10b53458ad gst-libs/gst/play/gstplay.c: When analyzing the pads of an element the bin is mostly in READY state so no caps were n...
Original commit message from CVS:
2004-01-12  Julien MOUTTE <julien@moutte.net>

* gst-libs/gst/play/gstplay.c: (gst_play_get_sink_element): When
analyzing the pads of an element the bin is mostly in READY state so
no caps were negotiated. This helper function needs to work with
_get_caps directly then. I was not freeing them though, added that to
fix the mem leak.
2004-01-12 16:48:48 +00:00
David Schleef
926ce51a28 Make sure everybody wraps #include "config.h" in #ifdef HAVE_CONFIG_H
Original commit message from CVS:
* ext/ffmpeg/gstffmpeg.c:
* ext/ffmpeg/gstffmpegcodecmap.c:
* ext/ffmpeg/gstffmpegdec.c:
* ext/ffmpeg/gstffmpegenc.c:
* ext/ffmpeg/gstffmpegprotocol.c:
* ext/gdk_pixbuf/gstgdkanimation.c:
* ext/jpeg/gstjpeg.c:
* ext/libpng/gstpng.c:
* ext/mpeg2dec/perftest.c:
* ext/speex/gstspeex.c:
* gst-libs/gst/resample/dtos.c:
* gst/intfloat/gstintfloatconvert.c:
* gst/oneton/gstoneton.c:
* gst/rtjpeg/RTjpeg.c:
* gst/rtp/gstrtp.c:
* sys/dxr3/dxr3init.c:
* sys/glsink/gstgl_nvimage.c:
* sys/glsink/gstgl_pdrimage.c:
* sys/glsink/gstglsink.c:
* testsuite/gst-lint:
Make sure everybody wraps #include "config.h" in #ifdef HAVE_CONFIG_H
2004-01-12 04:15:47 +00:00
David Schleef
37d7aecdb6 Remove all usage of gst_pad_get_caps(), and replace it with gst_pad_get_allowed_caps() or gst_pad_get_negotiated_cap().
Original commit message from CVS:
Remove all usage of gst_pad_get_caps(), and replace it with
gst_pad_get_allowed_caps() or gst_pad_get_negotiated_cap().
2004-01-12 03:40:18 +00:00
Benjamin Otte
929c5fff31 gst-libs/gst/tuner/tuner.*: GObjects aren't const.
Original commit message from CVS:
2004-01-12  Benjamin Otte  <in7y118@public.uni-hamburg.de>

* gst-libs/gst/tuner/tuner.c: (gst_tuner_class_init),
(gst_tuner_find_norm_by_name), (gst_v4l2_find_channel_by_name),
(gst_tuner_channel_changed), (gst_tuner_norm_changed),
(gst_tuner_frequency_changed), (gst_tuner_signal_changed):
* gst-libs/gst/tuner/tuner.h:
GObjects aren't const.
Add find_by_name functions.
Add checks to _changed functions.
* sys/v4l/gstv4ltuner.c: (gst_v4l_tuner_get_channel),
(gst_v4l_tuner_get_norm):
Fixes for above.
2004-01-12 02:11:03 +00:00
Benjamin Otte
883b6925fd gst-libs/gst/video/video.h: Fix caps template names to be understandable.
Original commit message from CVS:
2004-01-12  Benjamin Otte  <in7y118@public.uni-hamburg.de>

* gst-libs/gst/video/video.h:
Fix caps template names to be understandable.
Prefix everything with GST_VIDEO.
* ext/aalib/gstaasink.c:
* ext/divx/gstdivxdec.c:
* ext/divx/gstdivxenc.c:
* ext/gdk_pixbuf/gstgdkpixbuf.c:
* ext/hermes/gstcolorspace.c: (gst_colorspace_base_init):
* ext/jpeg/gstjpegdec.c: (raw_caps_factory):
* ext/jpeg/gstjpegenc.c: (raw_caps_factory):
* ext/libcaca/gstcacasink.c:
* ext/libpng/gstpngenc.c: (raw_caps_factory):
* ext/snapshot/gstsnapshot.c:
* ext/swfdec/gstswfdec.c:
* ext/xvid/gstxviddec.c:
* ext/xvid/gstxvidenc.c:
* gst/chart/gstchart.c:
* gst/deinterlace/gstdeinterlace.c:
* gst/effectv/gsteffectv.c:
* gst/flx/gstflxdec.c: (gst_flxdec_loop):
* gst/goom/gstgoom.c:
* gst/median/gstmedian.c:
* gst/monoscope/gstmonoscope.c: (gst_monoscope_init),
(gst_monoscope_srcconnect), (gst_monoscope_chain):
* gst/overlay/gstoverlay.c:
* gst/smooth/gstsmooth.c:
* gst/smpte/gstsmpte.c:
* gst/synaesthesia/gstsynaesthesia.c:
* gst/videocrop/gstvideocrop.c:
* gst/videodrop/gstvideodrop.c:
* gst/y4m/gsty4mencode.c:
* sys/qcam/gstqcamsrc.c:
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_palette_to_caps):
Make them work with new video.h file.
* sys/ximage/ximagesink.c: (gst_ximagesink_chain),
(gst_ximagesink_buffer_free), (gst_ximagesink_buffer_alloc):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_chain),
(gst_xvimagesink_buffer_free), (gst_xvimagesink_buffer_alloc):
Make it work with new buffer allocation system.
2004-01-12 02:01:52 +00:00
David Schleef
0cae73e234 ext/divx/gstdivxenc.c: remove bogus gst_caps_is_fixed() test
Original commit message from CVS:
* ext/divx/gstdivxenc.c: remove bogus gst_caps_is_fixed() test
* gst/debug/efence.c: (gst_efence_chain), (gst_fenced_buffer_new),
(gst_fenced_buffer_default_copy): Fix for rename of buffer private
structure members.
* gst/effectv/gstwarp.c: (gst_warptv_setup): Don't reset the time
value during a resize/renegotiation.
* gst/videofilter/gstvideofilter.c: (gst_videofilter_chain): use
gst_pad_alloc_buffer();
* sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_get),
(gst_v4lmjpegsrc_buffer_free): Fix for rename of buffer private
structure members.
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_get), (gst_v4lsrc_buffer_free):
Fix for rename of buffer private structure members.
* sys/ximage/ximagesink.c: (gst_ximagesink_chain),
(gst_ximagesink_buffer_free), (gst_ximagesink_buffer_alloc):
Fix for rename of buffer private structure members.
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_chain),
(gst_xvimagesink_buffer_free), (gst_xvimagesink_buffer_alloc):
Fix for rename of buffer private structure members.
2004-01-11 22:03:52 +00:00
Benjamin Otte
23bb1b8f35 I'm too lazy to comment this
Original commit message from CVS:
*** empty log message ***
2004-01-11 19:34:15 +00:00
Julien Moutte
7b89c1a8bf ext/ffmpeg/gstffmpegcolorspace.c: Implementing gst_pad_alloc_buffer to use optimized buffer allocation.
Original commit message from CVS:
* ext/ffmpeg/gstffmpegcolorspace.c: (gst_ffmpegcsp_chain):
Implementing gst_pad_alloc_buffer to use optimized buffer allocation.
* gst-libs/gst/xoverlay/xoverlay.c:
(gst_x_overlay_got_desired_size): Updating doc for the xid being 0.
* gst/videoscale/gstvideoscale.c: (gst_videoscale_chain):
Implementing gst_pad_alloc_buffer to use optimized buffer allocation.
* gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_get):
Implementing gst_pad_alloc_buffer to use optimized buffer allocation.
* sys/ximage/ximagesink.c: (gst_ximagesink_chain),
(gst_ximagesink_buffer_free), (gst_ximagesink_buffer_alloc),
(gst_ximagesink_set_xwindow_id), (gst_ximagesink_init): Implementing
the bufferalloc_function to replace bufferpools, fixing the XOverlay
interface implementation to handle xid being 0 and fix some bugs
triggered by Benjamin's testcase.
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_chain),
(gst_xvimagesink_buffer_free), (gst_xvimagesink_buffer_alloc),
(gst_xvimagesink_set_xwindow_id), (gst_xvimagesink_init): Implementing
the bufferalloc_function to replace bufferpools, fixing the XOverlay
interface implementation to handle xid being 0 and fix some bugs
triggered by Benjamin's testcase.
2004-01-09 18:05:57 +00:00
Ronald S. Bultje
ba219974c7 ext/divx/gstdivxenc.c: Use explicit caps - fix capsnego.
Original commit message from CVS:
2004-01-09  Ronald Bultje  <rbultje@ronald.bitfreak.net>

* ext/divx/gstdivxenc.c: (gst_divxenc_init):
Use explicit caps - fix capsnego.
* ext/xvid/gstxviddec.c:
* ext/xvid/gstxvidenc.c:
Remove macro-inside-macro which caused compile errors.
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_header):
Error out if it's not a RIFF file. Else we error out without
gst_element_error() which is not good...
2004-01-09 09:48:43 +00:00
David Schleef
899b633288 ext/ffmpeg/gstffmpegenc.c: Fix pad_link function to handle formats that ffmpeg returns as multiple caps structures.
Original commit message from CVS:
* ext/ffmpeg/gstffmpegenc.c: (gst_ffmpegenc_connect):
Fix pad_link function to handle formats that ffmpeg returns
as multiple caps structures.
* gst/videofilter/gstvideofilter.c: (gst_videofilter_chain):
Only complain if source buffer is _smaller_ than expected.
* gst/videoscale/gstvideoscale.c: (gst_videoscale_init),
(gst_videoscale_handle_src_event): Resize navigation events
when passing them upstream.
* gst/videotestsrc/gstvideotestsrc.c:
* gst/videotestsrc/gstvideotestsrc.h:
* gst/videotestsrc/videotestsrc.c:
* gst/videotestsrc/videotestsrc.h:
Rewrite many of the buffer painting functions to handle odd
sizes (for many formats, size%4!=0 or size%8!=0).  Most have
been verified to work with my video card.
* testsuite/gst-lint:  Add check for elements calling
gst_pad_get_caps() instead of gst_pad_get_allowed_caps().
2004-01-09 01:53:31 +00:00
Julien Moutte
2a35e3393f examples/gstplay/Makefile.am: Adding the interface library.
Original commit message from CVS:
* examples/gstplay/Makefile.am: Adding the interface library.
* gst-libs/gst/play/Makefile.am: Adding the interface library.
* gst-libs/gst/play/gstplay.c: (gst_play_set_video_sink): Connecting tothe XOverlay size signal instead of GstVideoSink.
* gst-libs/gst/play/gstplay.h: Including the XOverlay interface to check        GST_IS_X_OVERLAY before signal connect.
* gst-libs/gst/video/gstvideosink.c: (gst_videosink_class_init):
Removing the have_video_size signal.
* gst-libs/gst/video/gstvideosink.h: Removing the have_video_size signal        and associated public method.
* sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents),
(gst_ximagesink_sinkconnect): Using XOverlay public method to fire size
signal.
* sys/xvideo/xvideosink.c: (gst_xvideosink_sinkconnect),
(gst_xvideosink_xwindow_new): Using XOverlay public method to fire size
signal.
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_sinkconnect):
Using XOverlay public method to fire size signal.
2004-01-07 21:49:25 +00:00
Jan Schmidt
b22d0599b3 ext/dv/gstdvdec.c: Fix caps negotiation.
Original commit message from CVS:
* ext/dv/gstdvdec.c: (gst_dvdec_loop):
Fix caps negotiation.

* ext/dvdnav/dvdnavsrc.c: (dvdnavsrc_class_init),
(dvdnavsrc_update_buttoninfo), (dvdnavsrc_get),
(dvdnavsrc_get_event_mask), (dvdnav_handle_navigation_event),
(dvdnavsrc_event):
* ext/mpeg2dec/gstmpeg2dec.c:
* gst-libs/gst/navigation/navigation.c:
(gst_navigation_send_key_event), (gst_navigation_send_mouse_event):
* gst-libs/gst/navigation/navigation.h:
* gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_handle_src_event):
* sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
Super-simple first version of mouse and keyboard events. Clicking
on a DVD menu now works, although it may not take you where you
expected.

* gst/sine/gstsinesrc.c: (gst_sinesrc_src_fixate):
* gst/videotestsrc/gstvideotestsrc.c:
(gst_videotestsrc_src_fixate):
These fixate functions were broken - they never actually
fixated :)
2004-01-06 16:42:10 +00:00
Ronald S. Bultje
491381bf8c ext/shout/gstshout.c: fix for new caps system.
Original commit message from CVS:
2004-01-06  Ronald Bultje  <rbultje@ronald.bitfreak.net>

* ext/shout/gstshout.c: (gst_icecastsend_base_init),
(gst_icecastsend_init):
fix for new caps system.
* gst-libs/gst/mixer/mixertrack.h:
* sys/oss/gstossmixer.c: (gst_ossmixer_build_list):
Add 'master track' flag (for tools like ACME that only want to
change the main volume).
2004-01-06 15:59:02 +00:00
Benjamin Otte
5e15a670e8 gst-libs/gst/xoverlay/xoverlay.*: Add optional "desired size" signal and querying.
Original commit message from CVS:
2004-01-04  Benjamin Otte  <in7y118@public.uni-hamburg.de>

* gst-libs/gst/xoverlay/xoverlay.c: (gst_x_overlay_base_init),
(gst_x_overlay_got_xwindow_id), (gst_x_overlay_get_desired_size),
(gst_x_overlay_got_desired_size):
* gst-libs/gst/xoverlay/xoverlay.h:
Add optional "desired size" signal and querying.
2004-01-04 18:02:30 +00:00
David Schleef
84dc963101 Negotiation fixes
Original commit message from CVS:
Negotiation fixes
2004-01-02 07:25:10 +00:00
Ronald S. Bultje
274b241a38 configure.ac: Fix configure check for mpeg2enc. We need 1.6.1.93 instead of 1.6.1.92, since the pkg-config file of 1....
Original commit message from CVS:
2004-01-01  Ronald Bultje  <rbultje@ronald.bitfreak.net>

* configure.ac:
Fix configure check for mpeg2enc. We need 1.6.1.93 instead of
1.6.1.92, since the pkg-config file of 1.6.1.92 is borked and
it therefore uses the wrong include paths. Too bad... Note
that 1.6.1.93 is not release yet. ;).
Also add a check for mplex, which is now using the lib'ified
mplex from mjpegtools, too.
* ext/ffmpeg/gstffmpegcodecmap.c:
Add codec_tag for 3ivx/xvid. For xvid, this should fix playback
issues. I don't think ffmpeg handles 3ivx correctly, so this
probably won't work. But it won't hurt either.
* ext/ffmpeg/gstffmpegdec.c: (gst_ffmpegdec_connect),
(gst_ffmpegdec_chain):
* ext/ffmpeg/gstffmpegenc.c: (gst_ffmpegenc_connect),
(gst_ffmpegenc_chain_audio):
Fix memleak in audio encoding. Close codec if open fails, this
calls the cleanup routines so we can re-use the context.
* ext/mpeg2enc/gstmpeg2enc.cc:
Fix pad template names/types, fix memory issue with getcaps().
* ext/mpeg2enc/gstmpeg2encoder.cc:
* ext/mpeg2enc/gstmpeg2encoder.hh:
Fix compile issue with new caps system (const thingy).
* ext/mpeg2enc/gstmpeg2encpicturereader.cc:
* ext/mpeg2enc/gstmpeg2encpicturereader.hh:
We read a first frame right on initing, so that we have a caps
when we init the output. This caps is cached in padprivate and
read as first frame.
* ext/mplex/Makefile.am:
* ext/mplex/gstmplex.cc:
* ext/mplex/gstmplex.h:
* ext/mplex/gstmplex.hh:
* ext/mplex/gstmplexibitstream.cc:
* ext/mplex/gstmplexibitstream.hh:
* ext/mplex/gstmplexjob.cc:
* ext/mplex/gstmplexjob.hh:
* ext/mplex/gstmplexoutputstream.cc:
* ext/mplex/gstmplexoutputstream.hh:
We wrap mjpegtools mplex. So I rewrote the plugin. The old plugin
had issues, didn't do capsnego, supported only a subset of the
mplex features and required a mplex fork in our local CVS. Plus
that it worked agaist a very old mplex version. Rewriting was
faster than updating it.
* gst-libs/ext/Makefile.am:
* gst-libs/ext/mplex/INSTRUCT:
* gst-libs/ext/mplex/Makefile.am:
* gst-libs/ext/mplex/README:
* gst-libs/ext/mplex/TODO:
* gst-libs/ext/mplex/ac3strm_in.cc:
* gst-libs/ext/mplex/audiostrm.hh:
* gst-libs/ext/mplex/audiostrm_out.cc:
* gst-libs/ext/mplex/aunit.hh:
* gst-libs/ext/mplex/bits.cc:
* gst-libs/ext/mplex/bits.hh:
* gst-libs/ext/mplex/buffer.cc:
* gst-libs/ext/mplex/buffer.hh:
* gst-libs/ext/mplex/fastintfns.h:
* gst-libs/ext/mplex/format_codes.h:
* gst-libs/ext/mplex/inputstrm.cc:
* gst-libs/ext/mplex/inputstrm.hh:
* gst-libs/ext/mplex/lpcmstrm_in.cc:
* gst-libs/ext/mplex/mjpeg_logging.cc:
* gst-libs/ext/mplex/mjpeg_logging.h:
* gst-libs/ext/mplex/mjpeg_types.h:
* gst-libs/ext/mplex/mpastrm_in.cc:
* gst-libs/ext/mplex/mpegconsts.cc:
* gst-libs/ext/mplex/mpegconsts.h:
* gst-libs/ext/mplex/mplexconsts.hh:
* gst-libs/ext/mplex/multplex.cc:
* gst-libs/ext/mplex/outputstream.hh:
* gst-libs/ext/mplex/padstrm.cc:
* gst-libs/ext/mplex/padstrm.hh:
* gst-libs/ext/mplex/stillsstream.cc:
* gst-libs/ext/mplex/stillsstream.hh:
* gst-libs/ext/mplex/systems.cc:
* gst-libs/ext/mplex/systems.hh:
* gst-libs/ext/mplex/vector.cc:
* gst-libs/ext/mplex/vector.hh:
* gst-libs/ext/mplex/videostrm.hh:
* gst-libs/ext/mplex/videostrm_in.cc:
* gst-libs/ext/mplex/videostrm_out.cc:
* gst-libs/ext/mplex/yuv4mpeg.cc:
* gst-libs/ext/mplex/yuv4mpeg.h:
* gst-libs/ext/mplex/yuv4mpeg_intern.h:
* gst-libs/ext/mplex/yuv4mpeg_ratio.cc:
We don't fork mjpegtools' mplex in our CVS anymore.
* gst/avi/gstavidemux.c: (gst_avi_demux_src_getcaps),
(gst_avi_demux_add_stream):
* gst/avi/gstavidemux.h:
Add getcaps() function for proper caps nego. This makes some
parts of AVI playback/reading work.
* sys/ximage/ximagesink.c: (gst_ximagesink_sinkconnect):
Resize window on new capsnego. This is probably wrong, but
I'm still committing it because with current capsnego, the
first successfull capsnego is auto-fixated, therefore rounded
down to the lowest values in the caps. this results in a 16x16
XWindow that is not reized when real capsnego finishes.
Dave, I see more cases of this, do you know a proper solution?
* tools/gst-launch-ext.in:
Fix MPEG-4 AAC (Apple iPod/iTunes) file commandline.
2004-01-01 22:45:57 +00:00
David Schleef
e095323df1 Negotiation fixes.
Original commit message from CVS:
Negotiation fixes.
2003-12-31 08:02:04 +00:00
Jeremy Simon
e728ac80d1 Fix warnings
Original commit message from CVS:
Fix warnings
2003-12-27 19:37:17 +00:00
David Schleef
3b60021408 Merge CAPS branch
Original commit message from CVS:
Merge CAPS branch
2003-12-22 01:47:09 +00:00
Julien Moutte
6bb536ed65 examples/gstplay/player.c: using g_print instead of g_message.
Original commit message from CVS:
* examples/gstplay/player.c: (got_time_tick), (got_stream_length),
(got_video_size), (main): using g_print instead of g_message.
* gst-libs/gst/play/gstplay.c: (gst_play_pipeline_setup): Fixing EOS
signal which was not emitted because of "switch" element added to the
bin but not connected. (Removing from the bin temporarily)
2003-12-21 22:11:46 +00:00
Julien Moutte
99b2583820 configure.ac: X_DISPLAY_MISSING is set to 1 if AC_PATH_XTRA fails to find X development files. I don't understand the...
Original commit message from CVS:
* configure.ac: X_DISPLAY_MISSING is set to 1 if AC_PATH_XTRA fails to
find X development files. I don't understand the previous tests and
they fail on my debian/ppc unstable. This one works.
* examples/gstplay/player.c: (main): Set the pipeline to READY before
exiting.
* gst-libs/gst/play/gstplay.c: (gst_play_get_length_callback),
(gst_play_set_video_sink), (gst_play_set_audio_sink),
(gst_play_set_visualization): Add some safety checks in set_ methods
and state_change. This was throwing some ugly CRITICAL messages when
pipeline was getting disposed and casts were failing.
2003-12-21 19:48:40 +00:00
Ronald S. Bultje
ceb37e80dd Sorry Dave... Add mpegversion=1 to mp3 caps everywhere so that the autoplugger uses mad and not faad for mp3 decoding...
Original commit message from CVS:
Sorry Dave... Add mpegversion=1 to mp3 caps everywhere so that the autoplugger uses mad and not faad for mp3 decoding. This should fix mp3 playback.
2003-12-18 09:31:50 +00:00
Julien Moutte
b43a45315a Harmonize state of sinks with the pipeline state.
Original commit message from CVS:
Harmonize state of sinks with the pipeline state.
2003-12-16 22:52:01 +00:00
Ronald S. Bultje
2ffbe6345d MP3 is mpegversion=1, else it clashes with MPEG/AAC
Original commit message from CVS:
MP3 is mpegversion=1, else it clashes with MPEG/AAC
2003-12-15 08:39:52 +00:00
Thomas Vander Stichele
b1ed7faa30 dist fix
Original commit message from CVS:
dist fix
2003-12-14 21:27:49 +00:00
Thomas Vander Stichele
3e1b87c5ad distcheck fixes
Original commit message from CVS:
distcheck fixes
2003-12-14 19:50:00 +00:00
Julien Moutte
f1bd7a95af Synchronizing my changes.
Original commit message from CVS:
Synchronizing my changes.
2003-12-14 17:00:18 +00:00
Benjamin Otte
bf9d39d5f3 distcheck fixes: include built sources in CLEANFILES
Original commit message from CVS:
distcheck fixes: include built sources in CLEANFILES
2003-12-13 21:46:47 +00:00
Julien Moutte
6f1b63ea72 Using private structure for members so that we have more chance that the header is frozen. :)
Original commit message from CVS:
Using private structure for members so that we have more chance that the header is frozen. :)
2003-12-11 22:29:14 +00:00
David Schleef
03f79e151b Change GST_*_PADDING to _gst_private[GST_PADDING]
Original commit message from CVS:
Change GST_*_PADDING to _gst_private[GST_PADDING]
2003-12-09 10:02:08 +00:00
Benjamin Otte
046fa7602f make videofilter handle RGB correctly
Original commit message from CVS:
make videofilter handle RGB correctly
2003-12-09 01:17:28 +00:00
Julien Moutte
6bbaa16dcc Adding visualization.
Original commit message from CVS:
Adding visualization.
Removing interfaces.
Preparing videoswitch integration.
2003-12-08 16:01:49 +00:00
Ronald S. Bultje
da23b98e4c Whoops, this one is needed too
Original commit message from CVS:
Whoops, this one is needed too
2003-12-07 22:50:30 +00:00
Ronald S. Bultje
144a5cce24 Hm, darn, forgot to commit one dir
Original commit message from CVS:
Hm, darn, forgot to commit one dir
2003-12-07 22:49:04 +00:00
Ronald S. Bultje
00e390d267 Don't use class casts before they're initialized
Original commit message from CVS:
Don't use class casts before they're initialized
2003-12-07 22:39:03 +00:00
Ronald S. Bultje
ec6cfad7fd Typo
Original commit message from CVS:
Typo
2003-12-07 22:38:26 +00:00
Julien Moutte
bf0d2b97e2 time_tick works.
Original commit message from CVS:
time_tick works.
have_video_size works.
2003-12-07 19:02:31 +00:00
Julien Moutte
1ed00e22fd Updating with my local repository. It already plays video with sound :) Great !
Original commit message from CVS:
Updating with my local repository. It already plays video with sound :) Great !
2003-12-07 17:56:10 +00:00
Julien Moutte
9265dfe82b Adding my work in progress rewrite from scratch of libgstplay.
Original commit message from CVS:
Adding my work in progress rewrite from scratch of libgstplay.
2003-12-07 13:29:10 +00:00
Ronald S. Bultje
2ebfe048cc Move over from GstInterface to GstImplementsInterface. Also adds some signals to several interfaces
Original commit message from CVS:
Move over from GstInterface to GstImplementsInterface. Also adds some signals to several interfaces
2003-12-07 12:11:30 +00:00
Andy Wingo
80fece4f4b remove copyright field from plugins
Original commit message from CVS:
remove copyright field from plugins
2003-12-04 10:37:39 +00:00
Thomas Vander Stichele
27bca5bf8d fix X dependencies
Original commit message from CVS:
fix X dependencies
2003-12-03 12:49:29 +00:00
Thomas Vander Stichele
302786ba59 fixing up X test and general features test
Original commit message from CVS:
fixing up X test and general features test
2003-12-03 12:23:37 +00:00
Benjamin Otte
788acc4917 tagging stuff and build fixes. In detail:
Original commit message from CVS:
tagging stuff and build fixes. In detail:
- make gdk-pixbuf loader work when distchecking
- fix invalid syntax in ffmpeg Makefile. wildcards for EXTRA_DIST are not allowed. This broke builds where distdir != srcdir
- fix ffmpeg cvs grabbing when srcdir != distdir
- new id3tag plugin for id3 tag reading/writing (uses mad's libid3tag)
- mad and libid3tag require mad/libid3tag v0.15. Fixed configure to require that
- added ogg demuxer in ext/ogg. The demuxer does not handle events yet. Especially getting seeking right will require some effort or code copying from libvorbis.
- added raw vorbis detection to typefinding. oggdemux requires a typefind function to detect its contents.
- tags plugin in gst/tags. Provides API in <gst/tags/gsttagediting.h>. API includes tag matching GStreamer <=> ID3 and GStreamer <=> vorbis and writing/reading vorbiscomments or ID3v1 tags. Also included is a simple vorbiscomment reader/writer. Writing will not really work though until someone writes oggmux.
- various build fixes. Mostly missing (DIST)CLEANFILES.
- vorbisenc handles tag writing.

Now it's YOUR turn to fix and write more plugins that handle writing/reading of tags. :)
2003-11-24 04:08:48 +00:00
David Schleef
3863b258b2 distcheck fixes
Original commit message from CVS:
distcheck fixes
2003-11-22 04:06:50 +00:00
Julien Moutte
21269abc94 Fix a stupid mistake
Original commit message from CVS:
Fix a stupid mistake
2003-11-19 22:59:53 +00:00
Julien Moutte
1bbfb48f01 Removing the video_out things.
Original commit message from CVS:
Removing the video_out things.
2003-11-19 22:41:45 +00:00
Julien Moutte
ea49926624 Removing have_size as it is in GstVideoSink higher class.
Original commit message from CVS:
Removing have_size as it is in GstVideoSink higher class.
2003-11-19 22:40:59 +00:00
Julien Moutte
a58a759d9c Merged GstVideoOverlay features here.
Original commit message from CVS:
Merged GstVideoOverlay features here.
2003-11-17 16:29:38 +00:00
Julien Moutte
5499abfb9b Merging with GstXOverlay
Original commit message from CVS:
Merging with GstXOverlay
2003-11-17 15:42:32 +00:00
Leif Johnson
038f79929a + adding buffer-frames prop to standard caps macros
Original commit message from CVS:
+ adding buffer-frames prop to standard caps macros
2003-11-16 22:04:16 +00:00
Leif Johnson
736153ab06 + checking in plugin category changes
Original commit message from CVS:
+ checking in plugin category changes
2003-11-16 22:02:21 +00:00
Leif Johnson
96298704f0 + adding endianness conversion macros for floats
Original commit message from CVS:
+ adding endianness conversion macros for floats
2003-11-16 21:47:16 +00:00
Julien Moutte
b23dd4a720 Adding GstVideoOverlay interface.
Original commit message from CVS:
Adding GstVideoOverlay interface.
2003-11-15 15:02:49 +00:00
David Schleef
ac91fc230c er, no, signal should be VOID__POINTER. :)
Original commit message from CVS:
er, no, signal should be VOID__POINTER.  :)
2003-11-13 21:25:38 +00:00
David Schleef
3335546e2c signal should be VOID__VOID
Original commit message from CVS:
signal should be VOID__VOID
2003-11-13 01:48:53 +00:00
Benjamin Otte
fb844d50ec fix build system for X, XShm and Xv
Original commit message from CVS:
fix build system for X, XShm and Xv
2003-11-12 03:06:07 +00:00
Benjamin Otte
c0909a864e make colorspace handle the original RGB format, too, since it handles both. Add the template caps for the reverse for...
Original commit message from CVS:
make colorspace handle the original RGB format, too, since it handles both. Add the template caps for the reverse format to video.h
2003-11-11 22:40:43 +00:00
David Schleef
6f07363ea2 Change GstCaps to GstStructure in navigation events. Fix x[v]imagesink to scale navigation events.
Original commit message from CVS:
Change GstCaps to GstStructure in navigation events.  Fix x[v]imagesink
to scale navigation events.
2003-11-11 00:43:29 +00:00
David Schleef
6fe62bb933 remove gst_video_sink_push_ui_event()
Original commit message from CVS:
remove gst_video_sink_push_ui_event()
2003-11-10 22:48:31 +00:00
Christian Schaller
80afd3b002 disting fixes, pixbuf loader still borked in regards to disting
Original commit message from CVS:
disting fixes, pixbuf loader still borked in regards to disting
2003-11-09 22:48:58 +00:00
David Schleef
fc6ca363c3 remove pesky g_print()s
Original commit message from CVS:
remove pesky g_print()s
2003-11-08 03:02:14 +00:00
David Schleef
fc71bee2db much cleanup hacking made possible by plugin changes
Original commit message from CVS:
much cleanup hacking made possible by plugin changes
2003-11-08 02:52:41 +00:00
Jeremy Simon
3e99a8feb0 We don't need to patch wmadec.c and avcodec.h anymore
Original commit message from CVS:
We don't need to patch wmadec.c and avcodec.h anymore
2003-11-07 20:01:02 +00:00
Ronald S. Bultje
3f9d4cbd08 Remove all config.h includes from header files, add it to each source file and remove duplicate config.h includes fro...
Original commit message from CVS:
Remove all config.h includes from header files, add it to each source file and remove duplicate config.h includes from several source files
2003-11-07 12:47:02 +00:00
David Schleef
8d93616102 Don't set the actual video size in gst_video_sink_got_video_size().
Original commit message from CVS:
Don't set the actual video size in gst_video_sink_got_video_size().
This could probably be replaced once we handle pixel aspect ratios.
2003-11-07 04:36:41 +00:00
David I. Lehn
313fd06fa3 use int64 for frame-time
Original commit message from CVS:
use int64 for frame-time
2003-11-07 03:01:21 +00:00
Jeremy Simon
2addec025a Patch avcodec.h and wmadec.c to use flag1 and flag2 from context
Original commit message from CVS:
Patch avcodec.h and wmadec.c to use flag1 and flag2 from context
The asfdemux calculates the 2 values and gives them to the decoder via the pad/caps
The ffdec_wmav1 & ffdec_wmav2 put them in the ffmpeg context before they start to decode the data
2003-11-05 23:06:27 +00:00
Ronald S. Bultje
97beb82173 Some prettification fixes to this interface... Make it look good, make it simple and make it more GObject'ific. This ...
Original commit message from CVS:
Some prettification fixes to this interface... Make it look good, make it simple and make it more GObject'ific. This means that it now works largely almost the same as GObject (concerning arguments that you give to the functions, and layout of the virtual functions). I've also got an OSS implementation for this, but it needs one small change before I commit that. Expect that this weekend or so.
2003-11-05 22:59:46 +00:00
Ronald S. Bultje
3de18c044f Remove trailing slash in last line of comment, it causes issues with certain versions of automake
Original commit message from CVS:
Remove trailing slash in last line of comment, it causes issues with certain versions of automake
2003-11-05 10:57:58 +00:00
David Schleef
b81262ad9c Fix endianness issues manifested in pngenc. Fixes #125932
Original commit message from CVS:
Fix endianness issues manifested in pngenc.  Fixes #125932
2003-11-05 03:59:53 +00:00
Benjamin Otte
97197b70bd disable Altivec until someone fixes the build to allow it conditionally
Original commit message from CVS:
disable Altivec until someone fixes the build to allow it conditionally
2003-11-04 15:22:42 +00:00
David Schleef
94e0acef5b use top-level defined plugindir
Original commit message from CVS:
use top-level defined plugindir
2003-11-03 20:18:14 +00:00
David Schleef
f826f0f254 change libgst-interfaces to libgstinterfaces
Original commit message from CVS:
change libgst-interfaces to libgstinterfaces
2003-11-03 20:17:40 +00:00
Ronald S. Bultje
6fc1aed01c Bla
Original commit message from CVS:
Bla
2003-11-02 22:54:47 +00:00
Benjamin Otte
92ba809c91 first bunch of conversions to new plugin_init. Includes libs/gst, gst/id3, sys/oss, ext/gnomevfs, gst/typefind and ex...
Original commit message from CVS:
first bunch of conversions to new plugin_init. Includes libs/gst, gst/id3, sys/oss, ext/gnomevfs, gst/typefind and ext/mad.
You guessed it, everything Rhythmbox needs ;)
fixed BMP typefind and made gnomevfs one plugin instead of two while doing this
2003-10-31 20:03:29 +00:00
David Schleef
e206183335 Oops, didn't mean to remove riff
Original commit message from CVS:
Oops, didn't mean to remove riff
2003-10-31 20:01:39 +00:00
David Schleef
e94041f8ba minor cosmetic fix
Original commit message from CVS:
minor cosmetic fix
2003-10-30 22:05:24 +00:00
David Schleef
9b8d71fae8 Fix makefiles for GstPropertyProbe. Combine all interfaces into a single libgst-interface.
Original commit message from CVS:
Fix makefiles for GstPropertyProbe.  Combine all interfaces into
a single libgst-interface.
2003-10-30 02:36:42 +00:00
David Schleef
06fdc56290 First attempt (actually third) at a property probing interface
Original commit message from CVS:
First attempt (actually third) at a property probing interface
2003-10-30 02:32:00 +00:00
David Schleef
913ced9f7d change Iface to Interface
Original commit message from CVS:
change Iface to Interface
2003-10-30 01:43:45 +00:00
David Schleef
2665c843bd Change Class to Iface. Fix casting macros.
Original commit message from CVS:
Change Class to Iface.  Fix casting macros.
2003-10-29 05:12:18 +00:00
David Schleef
288210149e Don't use GstInterface
Original commit message from CVS:
Don't use GstInterface
2003-10-28 23:22:31 +00:00
Ronald S. Bultje
c488665ef7 Disable lame in ffmpeg, too
Original commit message from CVS:
Disable lame in ffmpeg, too
2003-10-28 23:02:38 +00:00
Benjamin Otte
112b1959da merge TYPEFIND branch. Major changes:
Original commit message from CVS:
merge TYPEFIND branch. Major changes:
- totally reworked type(find) system
- all typefind functions are in gst/typefind now
- more typefind functions then before
- some plugins might fail to compile now because I don't have them installed and they
a) require bytestream or
b) haven't had their typefind fixed.
Please fix those plugins and put the typefind functions into gst/typefind if they don't have dependencies
2003-10-28 20:52:41 +00:00
Ronald S. Bultje
6b58a5e865 Disable ogg/vorbis build inside ffmpeg. We didn't enable it anyway, and we've got our own ogg/vorbis codec already
Original commit message from CVS:
Disable ogg/vorbis build inside ffmpeg. We didn't enable it anyway, and we've got our own ogg/vorbis codec already
2003-10-28 12:20:52 +00:00
Ronald S. Bultje
f3d191297a Update FFMPEG snapshot to this morning's. Asked for by apoc.
Original commit message from CVS:
Update FFMPEG snapshot to this morning's. Asked for by apoc.
2003-10-26 11:51:17 +00:00
Ronald S. Bultje
7b80140b94 Gerd approved license change, so this code is now LGPL too
Original commit message from CVS:
Gerd approved license change, so this code is now LGPL too
2003-10-20 11:22:42 +00:00
Iain Holmes
cec071743c Add audio/x-id3 as a mimetype handled by mad.
Original commit message from CVS:
Add audio/x-id3 as a mimetype handled by mad.
Could also be a flac file apparently, but this allows it to work more times than it does currently
2003-10-13 20:53:53 +00:00
Ronald S. Bultje
0f124a569d Some interface implementations for video4linux/video4linux2 plugins: a Tuner interface, with which one can select inp...
Original commit message from CVS:
Some interface implementations for video4linux/video4linux2 plugins:
* a Tuner interface, with which one can select inputs and stations. Audio
work is underway here, but unfinished.
* A Xoverlay interface with which one can do simple overlay. Similar to
the API of the v4l/Xv XFree86 extension. Widget implementation for
GTK-2.0 coming up in the sandbox.
* Colorbalance - for adapting colors (brightness, contrast, etc.) - pretty
basic and maybe somewhat overdesigned. But it'll do for now.

Apart from these interfaces, there's also a loadable library 'xwindowlistener'
that listenes to X for the movement of a window and the overlap of other
windows. This is partly copied from xawtv (and thus partly GPL :(), but it's
needed for the xoverlay interface implementation in the v4l/v4l2 elements.

Lastly, some small changes to remove redundant properties from the v4l/v4l2
elements since these can be done much simpler. Comments appreciated!
2003-10-10 12:47:41 +00:00
Ronald S. Bultje
760a841261 Make GstMixerTrack a GObject. I also want to make it emit several signals, starting work is in here but it's not full...
Original commit message from CVS:
Make GstMixerTrack a GObject. I also want to make it emit several signals, starting work is in here but it's not fully implemented yet. for OSS, this will cause issues, but for ALSA, this is all automated.
2003-10-10 12:24:20 +00:00
Leif Johnson
6aa17f5bc1 changing 'channel' to 'track' to avoid naming overlap
Original commit message from CVS:
changing 'channel' to 'track' to avoid naming overlap
2003-10-09 22:02:35 +00:00
Leif Johnson
4f2d330fdf adding G_BEGIN and G_END_DECLS
Original commit message from CVS:
adding G_BEGIN and G_END_DECLS
2003-10-09 21:59:03 +00:00
David Schleef
c85a3d73a9 add object and struct padding
Original commit message from CVS:
add object and struct padding
2003-10-09 01:57:54 +00:00
Andy Wingo
dc35dbb595 /GstBuffer/GstData/ in the API where you can pass events. Fix the plugins to deal with that. Fixes #113488.
Original commit message from CVS:
/GstBuffer/GstData/ in the API where you can pass events. Fix the plugins to deal with that. Fixes #113488.
2003-10-08 16:08:22 +00:00
Julien Moutte
805130e62c There was a floating doubt on the way visualization is written. We use 'z' everywhere now. Both in gst-player and lib...
Original commit message from CVS:
There was a floating doubt on the way visualization is written. We use 'z' everywhere now. Both in gst-player and libgstplay
2003-10-06 10:29:27 +00:00
Brian Cameron
dce62eff63 Corrected macro in medi-info-priv.h so that it follows ISO standards, and removed broken return from void function in...
Original commit message from CVS:
Corrected macro in medi-info-priv.h so that it follows ISO standards,
and removed broken return from void function in navigation.c
2003-10-02 20:12:33 +00:00
Benjamin Otte
4258cd6910 make ffmpeg erroring out work:
Original commit message from CVS:
make ffmpeg erroring out work:
- don't built gst-libs/ext/ffmpeg
- don't update Tag when cvs command fails
2003-09-27 16:44:53 +00:00
David Schleef
05fe143049 add navigation subdir
Original commit message from CVS:
add navigation subdir
2003-09-27 03:17:34 +00:00
David Schleef
4fe6f5a34e add navigation interface
Original commit message from CVS:
add navigation interface
2003-09-27 03:16:28 +00:00
Iain Holmes
f600f3e4e8 Sync'ed files up to nautilus-media/media-info versions, and fixed them for errors and deprecated functions
Original commit message from CVS:
Sync'ed files up to nautilus-media/media-info versions, and fixed them for errors and deprecated functions
2003-09-23 22:49:35 +00:00
Julien Moutte
6d4f3a055b Changing have_xid to have_video_out to match the new GstVideoSink parent object.
Original commit message from CVS:
Changing have_xid to have_video_out to match the new GstVideoSink parent object.
Adding have_vis_size.
Fixing gst_play_connect_visualisation
2003-09-15 09:29:06 +00:00
Julien Moutte
a0453f999b Adding GstVideoSink object which will be used for video sink's subclassing.
Original commit message from CVS:
Adding GstVideoSink object which will be used for video sink's subclassing.
2003-09-14 12:21:06 +00:00
David Schleef
48b6f2010e Change some g_prints to GST_DEBUG
Original commit message from CVS:
Change some g_prints to GST_DEBUG
2003-09-14 11:50:14 +00:00
Ronald S. Bultje
a3890ba90c Copyright fix (code=mine)
Original commit message from CVS:
Copyright fix (code=mine)
2003-09-13 13:42:08 +00:00
Christian Schaller
e8dd975118 latest fixes to get everything disting and building like a charm
Original commit message from CVS:
latest fixes to get everything disting and building like a charm
2003-09-13 10:25:22 +00:00
David Schleef
dd772dd228 Add missing \
Original commit message from CVS:
Add missing \
2003-09-13 10:03:05 +00:00
Ronald S. Bultje
beea17989f Interface example: a mixer
Original commit message from CVS:
Interface example: a mixer
2003-09-13 01:20:04 +00:00
Christian Schaller
35989ae8d4 build and disting fixes
Original commit message from CVS:
build and disting fixes
2003-09-12 12:56:55 +00:00
David Schleef
ebb066d861 Remove all locally-defined optimization flags
Original commit message from CVS:
Remove all locally-defined optimization flags
2003-09-11 19:58:00 +00:00
David Schleef
83d6acc9de Add some stuff to disable bad asm code
Original commit message from CVS:
Add some stuff to disable bad asm code
2003-09-11 19:38:04 +00:00
David Schleef
2d5c283e8c Build ffmpeg using automake instead of ffmpeg's (broken) Makefiles
Original commit message from CVS:


Build ffmpeg using automake instead of ffmpeg's (broken) Makefiles
2003-09-11 19:28:09 +00:00
Christian Schaller
a7307dbb82 add cvsignores
Original commit message from CVS:
add cvsignores
2003-09-11 19:17:01 +00:00
Christian Schaller
f8a91d8492 misc minor build and rpm fixes
Original commit message from CVS:
misc minor build and rpm fixes
2003-09-11 19:02:00 +00:00
David Schleef
18731001f6 clean up varargs macro definition
Original commit message from CVS:
clean up varargs macro definition
2003-09-03 03:55:53 +00:00
David Schleef
ea50e56520 Remove vestigal code (#121174)
Original commit message from CVS:
Remove vestigal code (#121174)
2003-09-03 03:49:32 +00:00
Andrew Turner
905b39124f compilation problems on FreeBSD.
Original commit message from CVS:
Patch from Andrew Turner <zxombie@hotpop.com> (#120307).  Fixes
compilation problems on FreeBSD.
2003-08-21 04:42:56 +00:00
Christian Schaller
b5c527d9d8 needed fixes to enable building of RPMS of gst-plugins
Original commit message from CVS:
needed fixes to enable building of RPMS of gst-plugins
2003-08-19 21:05:26 +00:00
Ronald S. Bultje
45fd9bbd76 so bug cannot be closed yet
Original commit message from CVS:
Several solaris fixes, see #118473... One issue remaining (see the bug), so bug cannot be closed yet
2003-08-13 21:09:20 +00:00
Thomas Vander Stichele
6c10c45d11 dist fix
Original commit message from CVS:
dist fix
2003-08-13 12:18:42 +00:00
David Schleef
426cf5e9b3 clean up code, fix typos in lst checkin
Original commit message from CVS:
clean up code, fix typos in lst checkin
2003-08-13 06:43:01 +00:00
David Schleef
066eb23664 Move some code from wavparse
Original commit message from CVS:
Move some code from wavparse
2003-08-13 04:47:30 +00:00
David Schleef
bd2035d858 Make mplex build dependent on USE_MPLEX
Original commit message from CVS:
Make mplex build dependent on USE_MPLEX
2003-07-27 05:32:03 +00:00
David Schleef
67c9972ccf Add mplex subdir
Original commit message from CVS:
Add mplex subdir
2003-07-27 05:31:30 +00:00
David Schleef
63294c0655 Moved from gst-plugins/ext/mplex/. See that directory for older changelogs.
Original commit message from CVS:
Moved from gst-plugins/ext/mplex/.  See that directory for older
changelogs.
2003-07-26 03:01:58 +00:00
Brian Cameron
0288bc09de Adding needed license information. The patch was reviewed and approved by
Original commit message from CVS:
Adding needed license information.  The patch was reviewed and approved by
Christian Shaller.  Ronald Bultje and Benjamin also responded with
comments.
2003-07-25 10:50:48 +00:00
David Schleef
900db31d43 Force MAKEFLAGS to -j1 since ffmpeg is not -j safe
Original commit message from CVS:
Force MAKEFLAGS to -j1 since ffmpeg is not -j safe
2003-07-22 23:13:45 +00:00
David Schleef
45170036ad Remove .pc files from CLEANFILES, since it's done automatically by automake during distclean instead of (incorrectly)...
Original commit message from CVS:
Remove .pc files from CLEANFILES, since it's done automatically by
automake during distclean instead of (incorrectly) during clean.
2003-07-22 21:24:51 +00:00
Joshua N. Pritikin
da57aaa48e Fix another instance of audio/raw -> audio/x-raw-int and video/raw -> video/x-raw-{yuv,rgb}. There probably should b...
Original commit message from CVS:
Fix another instance of audio/raw -> audio/x-raw-int and
video/raw -> video/x-raw-{yuv,rgb}.  There probably should
be a audio/x-raw-float for jacksink but I didn't add that.
2003-07-22 09:09:27 +00:00
Leif Johnson
abe6f76b9f + the last of the float caps changes ... these are a bit more pervasive
Original commit message from CVS:
+ the last of the float caps changes ... these are a bit more pervasive
2003-07-19 23:47:42 +00:00
Andy Wingo
7c6f49bf6b actually recurse into sndfile if we are able big ladspa cleanups, mainly to comply with the buffer-frames caps proper...
Original commit message from CVS:
* actually recurse into sndfile if we are able
* big ladspa cleanups, mainly to comply with the buffer-frames caps property, but also general
cleanups
- the samplerate prop is gone, if you want to set it explicitly (as in for get-based plugins)
you need to use a filtered connection, just like with buffer-frames
* big float2int and int2float changes for buffer-frames compatibility - I think it's quite a bit
simpler
* make the ossclock general, add it to gstaudio, and use it in sndfile as well

i need to update mimetypes, but that's coming soon. there are some other plugins that don't
support buffer-frames, i guess i need to get around to fixing them as well.
2003-07-16 16:08:13 +00:00
David Schleef
64b6931e61 Caps fixes and general cleanup.
Original commit message from CVS:
Caps fixes and general cleanup.
2003-07-14 08:32:38 +00:00
Julien Moutte
06ae4f4057 fixing buggy error handling
Original commit message from CVS:
fixing buggy error handling
2003-07-12 09:18:31 +00:00
Christian Schaller
a88415b831 A big rewrite of the spec file and some ffmpeg disting fixes. The ffmpeg plugin still doesn't build for me, but I thi...
Original commit message from CVS:
A big rewrite of the spec file and some ffmpeg disting fixes. The ffmpeg plugin still doesn't build for me, but I think I am getting close :)
2003-07-07 22:40:55 +00:00
Ronald S. Bultje
b005531324 New mimetypes gone into effect today - this commit changes all old mimetypes over to the new mimetypes spec as descri...
Original commit message from CVS:
New mimetypes gone into effect today - this commit changes all old mimetypes over to the new mimetypes spec as described in the previous commit's document. Note: some plugins will break, some pipelines will break, expect HEAD to be broken or at least not 100% working for a few days, but don't forget to report bugs
2003-07-06 20:49:52 +00:00
David Schleef
374d12d95b Add a base init function
Original commit message from CVS:
Add a base init function
2003-07-02 08:08:34 +00:00
David Schleef
8bbfd0c011 GstVideofilter abstract class for video filters
Original commit message from CVS:
GstVideofilter abstract class for video filters
2003-07-02 05:22:47 +00:00
Benjamin Otte
ffa5706370 compatibility fix for new GST_DEBUG stuff.
Original commit message from CVS:
compatibility fix for new GST_DEBUG stuff.
Includes fixes for missing includes for config.h and unistd.h

I only ensured for plugins I can build that they work, so if some of them are still broken, you gotta fix them yourselves unfortunately.
2003-06-29 19:46:12 +00:00
Ronald S. Bultje
2913999cbe New ffmpeg version fixes mpeg decoding segfaults (mpeg1)
Original commit message from CVS:
New ffmpeg version fixes mpeg decoding segfaults (mpeg1)
2003-06-17 11:43:24 +00:00
Julien Moutte
8860bd5ac5 Fixing indentation
Original commit message from CVS:
Fixing indentation
2003-06-16 10:22:39 +00:00
Benjamin Otte
5ee54c03c0 gcc 3.3 and ppc fixes
Original commit message from CVS:
gcc 3.3 and ppc fixes
2003-06-14 20:10:13 +00:00
Ronald S. Bultje
9b647bc79b Fix 114948 - adds some autotools magic that makes sure that we only make install ffmpeg if its actually there
Original commit message from CVS:
Fix 114948 - adds some autotools magic that makes sure that we only make install ffmpeg if its actually there
2003-06-11 19:42:20 +00:00
Thomas Vander Stichele
94d20d0e32 warn on state change failures
Original commit message from CVS:
warn on state change failures
2003-06-07 22:28:39 +00:00
Julien Moutte
6af1a5e382 Fixing error message when a plugin is missing (gst-player -> libgstplay).
Original commit message from CVS:
Fixing error message when a plugin is missing (gst-player -> libgstplay).
Pipeline state change on error is now done directly in error callback, there's still something weird with oss /dev/dsp ressource busy error.
2003-06-01 11:43:08 +00:00
Julien Moutte
2c32dd6f5a Error handling is inside and works :)
Original commit message from CVS:
Error handling is inside and works :)
Mutexes have been removed. That should fix UI responsiveness problems on query_length. Please heavily test to check if they were really needed.
Some fixes in GClosures
2003-05-30 22:36:19 +00:00
Ronald S. Bultje
812a8edbe5 Use new ffmpeg revision
Original commit message from CVS:
Use new ffmpeg revision
2003-05-29 19:33:58 +00:00
Thomas Vander Stichele
e6e64e33f2 don't warn on recoverable problems
Original commit message from CVS:
don't warn on recoverable problems
2003-05-29 17:00:55 +00:00
Wim Taymans
8e7b367524 UNITS -> DEFAULT
Original commit message from CVS:
UNITS -> DEFAULT
2003-05-24 10:41:58 +00:00
Brian Cameron
803c7ab378 This ISO_VARARGS macro is more consistant with the others.
Original commit message from CVS:
This ISO_VARARGS macro is more consistant with the others.
2003-05-22 10:03:31 +00:00
Brian Cameron
a27b4e441f Corrected the configure.ac so it actually works. Updated some c files so that they build on Solaris. This mostly in...
Original commit message from CVS:
Corrected the configure.ac so it actually works.  Updated some c files
so that they build on Solaris.  This mostly involved supporting ISO
style variable-argument macros.
2003-05-21 18:16:30 +00:00
Brian Cameron
361c9eff09 Updated autogen.sh/configure.ac and various Makefiles to make the configure script set up all gcc specific compiler a...
Original commit message from CVS:
Updated autogen.sh/configure.ac and various Makefiles to make the
configure script set up all gcc specific compiler arguments, rather
than hardcoding them in the Makefile.am files
2003-05-21 15:16:51 +00:00
David Schleef
346b918257 Links also against libgstcontrol
Original commit message from CVS:
Links also against libgstcontrol
2003-05-21 07:47:28 +00:00
Benjamin Otte
15f0448b79 make error signal use the right types
Original commit message from CVS:
make error signal use the right types
2003-05-12 16:31:31 +00:00
Benjamin Otte
2bc184b6ab uh, maybe putting new signals before LAST_SIGNAL might help against memory corruption?
Original commit message from CVS:
uh, maybe putting new signals before LAST_SIGNAL might help against memory corruption?
2003-05-12 16:25:26 +00:00
Julien Moutte
dffe04541c deep-notify can come from pads too, replacing GstElement by GstObject.
Original commit message from CVS:
deep-notify can come from pads too, replacing GstElement by GstObject.
2003-05-06 22:22:39 +00:00
Ronald S. Bultje
2c8acd1196 Gcc < 3.0 fix
Original commit message from CVS:
Gcc < 3.0 fix
2003-05-02 20:13:08 +00:00
Julien Moutte
550f0bd378 Make vis_video_thread play when connected afterwards
Original commit message from CVS:
Make vis_video_thread play when connected afterwards
This generates some segfaults in gst_thread but we need to fix that
2003-04-28 10:52:29 +00:00
Thomas Vander Stichele
4ffaf1f893 more uninstalled fixes
Original commit message from CVS:
more uninstalled fixes
2003-04-27 15:20:07 +00:00
Christian Schaller
232891aaf1 Removing avifile and http plugins, also fixing some disting issues
Original commit message from CVS:
Removing avifile and http plugins, also fixing some disting issues
2003-04-23 19:42:12 +00:00
Julien Moutte
7b2a8e25dc Made a theorical libgstplay which refs/unrefs elements before putting them in AsyncQueue.
Original commit message from CVS:
Made a theorical libgstplay which refs/unrefs elements before putting them in AsyncQueue.
Added a "pipeline_error" signal which will later allow the player and apps to detect that pipeline was unable to play and why...
This version is NOT STABLE AT ALL. it will need fixes in core but i commit it as is so that we fix those problems
2003-04-20 21:06:55 +00:00
Julien Moutte
7877428028 Changed sanity checks according to BBB feedback
Original commit message from CVS:
Changed sanity checks according to BBB feedback
2003-04-17 10:20:25 +00:00
Julien Moutte
7ed0931911 make Company happy : Changed visualisation pipeline structure.. audio sink is directly connected to tee so no queue b...
Original commit message from CVS:
make Company happy : Changed visualisation pipeline structure.. audio sink is directly connected to tee so no queue between volume and audio sink...
2003-04-13 20:02:34 +00:00
Julien Moutte
83d1b34deb Adding a lot of sanity checks
Original commit message from CVS:
Adding a lot of sanity checks
Checking if vis pad is connected before linking/unlinking
2003-04-10 13:20:02 +00:00
Ronald S. Bultje
f3d3912f45 Port some changes from 0.6.0/0.6.1 back to HEAD (.pc files moved to pkg-config/, .m4 file moved from plugins to core)
Original commit message from CVS:
Port some changes from 0.6.0/0.6.1 back to HEAD (.pc files moved to pkg-config/, .m4 file moved from plugins to core)
2003-04-07 21:34:30 +00:00
David Schleef
3ec7ab1f08 Remove CVS/Tag from dist. Should go into 0.6.1.
Original commit message from CVS:
Remove CVS/Tag from dist.  Should go into 0.6.1.
2003-04-07 19:00:09 +00:00
Ronald S. Bultje
dec39b7775 add comments
Original commit message from CVS:
add comments
2003-04-05 21:16:02 +00:00
Ronald S. Bultje
996b613d22 Well, ok, one more patch - fix the well-known v4l compilation problems inside ffmpeg too
Original commit message from CVS:
Well, ok, one more patch - fix the well-known v4l compilation problems inside ffmpeg too
2003-04-04 21:17:10 +00:00
David Schleef
9dd19f1af6 Add $(GST_LIBS) to LIBADD for libgstgconf. Should go into 0.6.1.
Original commit message from CVS:
Add $(GST_LIBS) to LIBADD for libgstgconf.  Should go into 0.6.1.
2003-04-04 19:16:46 +00:00
Julien Moutte
7f41fdf9d2 Incorporating visualisation in libgstplay adding have_vis_xid signal and a new type of GstPlay object (VIDEO_VISUALIS...
Original commit message from CVS:
Incorporating visualisation in libgstplay
adding have_vis_xid signal and a new type of GstPlay object (VIDEO_VISUALISATION)
2003-04-02 21:33:12 +00:00
Julien Moutte
5208f05176 Fixed seeking discont with visualisation
Original commit message from CVS:
Fixed seeking discont with visualisation
2003-03-30 10:21:40 +00:00
Julien Moutte
757db2e087 Added gst_gconf function to render visualisation plugin from GConf key "default/visualisation"
Original commit message from CVS:
Added gst_gconf function to render visualisation plugin from GConf key "default/visualisation"
Enjoyed that commit to document those functions and remove commented functions
2003-03-29 19:48:38 +00:00
Benjamin Otte
02872d827d fix compile error
Original commit message from CVS:
fix compile error
2003-03-24 02:58:13 +00:00
Julien Moutte
efcf227524 gst_play_get_sink_element has been improved :
Original commit message from CVS:
gst_play_get_sink_element has been improved :
- Recursively searches in bins
- Able to find sink_element of a specified GstPlaySinkType (audio,video,any)

This now allow us to send very complex bins to GstPlay for audio/video sink without breaking length querying, eos signaling, etc..
2003-03-03 17:58:07 +00:00
Christian Schaller
d410130574 add FreeBSD patches from Andrew Turner and add missing ivorbis m4 to cvs, also disable ivorbis plugin as the test mis...
Original commit message from CVS:
add FreeBSD patches from Andrew Turner and add missing ivorbis m4 to cvs, also disable ivorbis plugin as the test mistakes ordinary vorbis for tremor
2003-02-23 19:58:50 +00:00
Julien Moutte
751072c948 Documented libgstplay's public methods
Original commit message from CVS:
Documented libgstplay's public methods
2003-02-07 11:02:21 +00:00
Julien Moutte
bcea95f6cb Code cleanup and reorganization
Original commit message from CVS:
Code cleanup and reorganization
I ll now be able to make nice gtk-doc for this lib
Fixed latency bug on async_queue pop
X video window now comes immediatly
2003-02-06 23:17:37 +00:00
David Schleef
0a1d9ddd6a Change %lld, %llu to G_G[U]INT64_FORMAT globally. Fix bad pointer->integer conversion.
Original commit message from CVS:
Change %lld, %llu to G_G[U]INT64_FORMAT globally.  Fix bad pointer->integer
conversion.
2003-02-02 05:26:29 +00:00
Owen Fraser-Green
60e053f79c Renamed GST_RIFF_WAVE_FORMAT_DIVX and _divx to _DIVX_WMAV1 and _DIVX_WMAV2 and added _WMAV9
Original commit message from CVS:
Renamed GST_RIFF_WAVE_FORMAT_DIVX and _divx to _DIVX_WMAV1 and _DIVX_WMAV2 and added _WMAV9
2003-01-31 23:06:18 +00:00
Ronald S. Bultje
8300e7ee8b Add metadata/streaminfo to AVI
Original commit message from CVS:
Add metadata/streaminfo to AVI
2003-01-31 19:37:25 +00:00
Ronald S. Bultje
d1cdccccce Add untested get-video-size function (bug 104360)
Original commit message from CVS:
Add untested get-video-size function (bug 104360)
2003-01-30 20:47:43 +00:00
Wim Taymans
a635592250 - Add strn tag
Original commit message from CVS:
- Add strn tag
- fix memleak in debugging code
2003-01-21 22:23:11 +00:00
Thomas Vander Stichele
0b921d3211 uninstall fix
Original commit message from CVS:
uninstall fix
2003-01-20 11:58:56 +00:00
Wim Taymans
7ad0c4a30f Hmm, forgot to commit this crucial define..
Original commit message from CVS:
Hmm, forgot to commit this crucial define..
2003-01-19 01:53:34 +00:00
Steve Baker
9f2240fa5a added func gst_play_set_data_src removed pipeline GST_PLAY_PIPE_VIDEO_THREADSAFE - it never worked and its just takin...
Original commit message from CVS:
added func gst_play_set_data_src
removed pipeline GST_PLAY_PIPE_VIDEO_THREADSAFE - it never worked and its just taking up space
2003-01-18 04:37:32 +00:00
Thomas Vander Stichele
87cf678fcc this is a better way
Original commit message from CVS:
this is a better way
2003-01-16 17:13:00 +00:00
Thomas Vander Stichele
c079cc492b patch ffmpeg
Original commit message from CVS:
patch ffmpeg
2003-01-16 16:58:29 +00:00
Thomas Vander Stichele
5f3ceac531 thaytan's ffmpeg function renaming patch
Original commit message from CVS:
thaytan's ffmpeg function renaming patch
2003-01-16 16:57:42 +00:00
Iain Holmes
6b00467cbb Add a new riff id tag: dmlh
Original commit message from CVS:
Add a new riff id tag: dmlh
2003-01-15 00:38:54 +00:00
Thomas Vander Stichele
ddce1dd0ec this should fix Uraeus's rpm making problem
Original commit message from CVS:
this should fix Uraeus's rpm making problem
2003-01-10 11:01:27 +00:00
Thomas Vander Stichele
9032557b33 another batch of connect->link fixes please let me know about issues and please refrain of making them yourself, so t...
Original commit message from CVS:
another batch of connect->link fixes
please let me know about issues
and please refrain of making them yourself, so that I don't spend double
the time resolving conflicts
2003-01-10 10:22:25 +00:00
Thomas Vander Stichele
346d1228b9 merge back from release
Original commit message from CVS:
merge back from release
2003-01-08 16:41:27 +00:00
Julien Moutte
7dc8ac4b51 Remove some unneeded comments
Original commit message from CVS:
Remove some unneeded comments
2003-01-07 19:13:41 +00:00
Julien Moutte
02e8bc5a99 Fixed gst_play_dispose function so that GstPlay objects can be unrefed safely.
Original commit message from CVS:
Fixed gst_play_dispose function so that GstPlay objects can be unrefed safely.
Optimized some g_idle_add for the signal poller.
2003-01-07 19:02:44 +00:00
Thomas Vander Stichele
3184990053 ignore ffmpeg dir
Original commit message from CVS:
ignore ffmpeg dir
2003-01-05 23:14:02 +00:00
Thomas Vander Stichele
cdf5c489b8 only slurp in cvs mode
Original commit message from CVS:
only slurp in cvs mode
2003-01-05 11:53:32 +00:00
Thomas Vander Stichele
fbbd5781ff a new way of handling ffmpeg cvs
Original commit message from CVS:
a new way of handling ffmpeg cvs
2003-01-05 11:31:01 +00:00
Thomas Vander Stichele
eb7e1bfc37 some dist fixes
Original commit message from CVS:
some dist fixes
2003-01-03 17:03:22 +00:00
David I. Lehn
d7e508a4f6 exciting .cvsignore updates!
Original commit message from CVS:
exciting .cvsignore updates!
2002-12-31 18:14:08 +00:00
Wim Taymans
12bd93773c Fix plugins for new query API
Original commit message from CVS:
Fix plugins for new query API
2002-12-30 17:53:18 +00:00
Thomas Vander Stichele
60f2fdc052 ffmpeg fix
Original commit message from CVS:
ffmpeg fix
2002-12-30 12:41:03 +00:00
Thomas Vander Stichele
80d5716d8e yes sir i can boogie
Original commit message from CVS:
yes sir i can boogie
2002-12-27 16:27:10 +00:00
Thomas Vander Stichele
02698e52e5 sigh
Original commit message from CVS:
sigh
2002-12-27 16:24:01 +00:00
Thomas Vander Stichele
4c87c9053a geez, me is teh suck
Original commit message from CVS:
geez, me is teh suck
2002-12-27 16:16:19 +00:00
Thomas Vander Stichele
62e05fd991 slurp in ffmpeg a) please test b) please help me figure out how to portably link in a .a file in our .so plugin
Original commit message from CVS:
slurp in ffmpeg

a) please test
b) please help me figure out how to portably link in a .a file in our
.so plugin
2002-12-26 10:53:21 +00:00
Christian Schaller
89953cc76c commit patch from Nall for riffutil
Original commit message from CVS:
commit patch from Nall for riffutil
2002-12-12 11:42:14 +00:00
Thomas Vander Stichele
a1045db325 fix 100982
Original commit message from CVS:
fix 100982
2002-12-11 23:32:52 +00:00
Thomas Vander Stichele
48d140aee2 merge back from release branch
Original commit message from CVS:
merge back from release branch
2002-12-11 21:58:36 +00:00
Steve Baker
6bbe57144b Applied state-saving patch from Leif Johnson
Original commit message from CVS:
Applied state-saving patch from Leif Johnson
2002-12-11 04:39:45 +00:00
Iain Holmes
3332daedad Replace audio/mp3 with audio/x-mp3 and audio/x-flac with application/x-flac
Original commit message from CVS:
Replace audio/mp3 with audio/x-mp3 and audio/x-flac with application/x-flac
2002-12-08 17:20:45 +00:00
Thomas Vander Stichele
3b540378fb parallel install fixes
Original commit message from CVS:
parallel install fixes
2002-12-08 14:50:09 +00:00
Steve Baker
0a8b5e07e3 go to READY if there is a gst_element_error
Original commit message from CVS:
go to READY if there is a gst_element_error
2002-12-01 01:55:45 +00:00
Thomas Vander Stichele
803632635c flac debug cleanup m4 for ffmpeg
Original commit message from CVS:
flac debug cleanup
m4 for ffmpeg
2002-11-26 14:51:34 +00:00
Thomas Vander Stichele
5b8cf51087 some fixes, suggestion by thaytan to make _get and _set work similar
Original commit message from CVS:
some fixes, suggestion by thaytan to make _get and _set work similar
2002-11-20 12:15:49 +00:00
Thomas Vander Stichele
ff8345157e adding media info
Original commit message from CVS:
adding media info
2002-11-11 13:59:51 +00:00
Thomas Vander Stichele
c892dfada8 media info lib
Original commit message from CVS:
media info lib
2002-11-11 13:59:16 +00:00
Thomas Vander Stichele
4b83831f48 muhaha
Original commit message from CVS:
muhaha
2002-11-09 18:31:27 +00:00
Thomas Vander Stichele
9bcaf77637 applied dolphy's patch
Original commit message from CVS:
applied dolphy's patch
2002-11-05 10:12:01 +00:00
Thomas Vander Stichele
114a380550 adding audio
Original commit message from CVS:
adding audio
2002-10-29 18:05:28 +00:00
Steve Baker
e9a8cd2599 libgstplay has a new home. it still needs to be packaged though
Original commit message from CVS:
libgstplay has a new home. it still needs to be packaged though
2002-10-27 20:59:41 +00:00
Ronald S. Bultje
0fd0914606 Add ogg/vorbis support for avi files
Original commit message from CVS:
Add ogg/vorbis support for avi files
2002-10-27 18:08:43 +00:00
Thomas Vander Stichele
0f75bf8d83 a no-brainer addition
Original commit message from CVS:
a no-brainer addition
2002-10-25 14:31:15 +00:00
Andy Wingo
db3abd8393 added functions gst_gconf_get_default_{audio,video}_sink -- this is so the defaults library (gstgconf) can be used by...
Original commit message from CVS:
added functions gst_gconf_get_default_{audio,video}_sink -- this is so the
defaults library (gstgconf) can be used by applications with uninstalled gstreamer
(ie, no gconf keys). More specifically, these functions enable the player to
work uninstalled, which is a good thing (tm).
2002-10-16 19:47:05 +00:00
David I. Lehn
cc12111d12 build as a lib not a plugin
Original commit message from CVS:
build as a lib not a plugin
2002-10-03 03:17:16 +00:00
Andy Wingo
1494ead81c %f is double
Original commit message from CVS:
%f is double
2002-09-29 17:16:40 +00:00
Ronald S. Bultje
895c730a13 This adds a generic video functions library
Original commit message from CVS:
This adds a generic video functions library
2002-09-26 12:20:12 +00:00
David Schleef
58f484799d Add config.h, since we use HAVE_CPU_PPC
Original commit message from CVS:
Add config.h, since we use HAVE_CPU_PPC
2002-09-15 21:39:54 +00:00
Wim Taymans
bda510b4a0 const fixes
Original commit message from CVS:
const fixes
2002-09-12 20:53:56 +00:00
Steve Baker
dacd65ae01 remove some debugging. it works already
Original commit message from CVS:
remove some debugging. it works already
2002-09-10 17:08:14 +00:00
Thomas Vander Stichele
4d5888a1f5 some simple fixes and additions
Original commit message from CVS:
some simple fixes and additions
2002-09-09 08:39:58 +00:00
Thomas Vander Stichele
6e8bd7473c oops, longtime pending fix
Original commit message from CVS:
oops, longtime pending fix
2002-08-06 23:55:08 +00:00
Steve Baker
4c137abcf6 one bin is enough
Original commit message from CVS:
one bin is enough
2002-07-28 15:06:06 +00:00
Thomas Vander Stichele
9392c00442 this is neater
Original commit message from CVS:
this is neater
2002-07-15 11:17:24 +00:00
Thomas Vander Stichele
386dfd7137 more of these
Original commit message from CVS:
more of these
2002-07-15 11:13:10 +00:00
Thomas Vander Stichele
390d18891e more useful info
Original commit message from CVS:
more useful info
2002-07-15 11:09:26 +00:00
Thomas Vander Stichele
7d83fadc7d avoid division by zero and add warning
Original commit message from CVS:
avoid division by zero and add warning
2002-07-15 11:06:33 +00:00
Thomas Vander Stichele
e7bb468bd8 small fixes
Original commit message from CVS:
small fixes
2002-07-12 21:09:17 +00:00
Thomas Vander Stichele
0577fa9b94 parse to BIN dumbass
Original commit message from CVS:
parse to BIN dumbass
2002-07-12 13:40:53 +00:00
Thomas Vander Stichele
f6c10c5913 api changes and render/ghost functions
Original commit message from CVS:
api changes and render/ghost functions
2002-07-12 09:41:25 +00:00
Thomas Vander Stichele
1015f47d6b merge from release branch
Original commit message from CVS:
merge from release branch
2002-07-09 10:45:37 +00:00
David Schleef
be67416805 Fix warning
Original commit message from CVS:
Fix warning
2002-07-02 03:29:21 +00:00
Thomas Vander Stichele
26019f6e28 gconf stuff
Original commit message from CVS:
gconf stuff
2002-07-01 15:39:58 +00:00
David Schleef
7f07809cd9 Fix buffer allocation problem.
Original commit message from CVS:
Fix buffer allocation problem.
2002-06-29 00:40:37 +00:00
Thomas Vander Stichele
61a065eabd header relocation
Original commit message from CVS:
header relocation
2002-06-22 23:37:20 +00:00
Andy Wingo
243cd11675 add float support to resampling lib clean up header file
Original commit message from CVS:
* add float support to resampling lib
* clean up header file
2002-05-29 06:13:26 +00:00
Andy Wingo
d734006e3c floatcast has no lib, it's just headers
Original commit message from CVS:
floatcast has no lib, it's just headers
2002-05-20 22:16:00 +00:00
Thomas Vander Stichele
726d1a035d assimilated.
Original commit message from CVS:
assimilated.
2002-05-13 14:34:11 +00:00
Wim Taymans
23ba333a82 Fixed a typo in the idct plugin name make gstriff a real plugin
Original commit message from CVS:
Fixed a typo in the idct plugin name
make gstriff a real plugin
2002-05-08 20:10:56 +00:00
Steve Baker
21702eee6e adds functions gst_cast_float and gst_cast_double. Will use lrint and lrintf if available and a pure c rounding macro...
Original commit message from CVS:
adds functions gst_cast_float and gst_cast_double. Will use lrint and lrintf if available and a pure c rounding macro if not.

This is where optimised asm alternatives can be placed - omega, this means you!

However you have to prove that your asm version is faster than lrintf ;)
2002-04-26 09:21:14 +00:00
Steve Baker
e9ab9a2b9d currently this creates an empty shared library, just so the header can be installed. I would appreciate help on how ...
Original commit message from CVS:
currently this creates an empty shared library, just so the header can be installed.  I would appreciate help on how to do this properly with automake.
2002-04-26 09:18:39 +00:00
Steve Baker
2a968ffb81 add dir for floatcast header
Original commit message from CVS:
add dir for floatcast header
2002-04-26 09:17:06 +00:00
Thomas Vander Stichele
7b694d63f8 spec fixes gst-launch-ext additions trying to fix up ieeetest but failed function collision in avifile dir
Original commit message from CVS:
* spec fixes
* gst-launch-ext additions
* trying to fix up ieeetest but failed
* function collision in avifile dir
2002-04-13 00:46:23 +00:00
Thomas Vander Stichele
b08ac04b97 comment them out
Original commit message from CVS:
comment them out
2002-04-12 21:10:31 +00:00
Thomas Vander Stichele
14c15e9e3c please someone fix the necessary pointer cast for this test to work
Original commit message from CVS:
please someone fix the necessary pointer cast for this test to work
2002-04-12 19:28:21 +00:00
Andy Wingo
f7ebc8f95d GstPadTemplate <-> gst_pad_template <-> GST_PAD_TEMPLATE same with *factory and typefind.
Original commit message from CVS:
GstPadTemplate <-> gst_pad_template <-> GST_PAD_TEMPLATE

same with *factory and typefind.

also, some -Werror fixes.
2002-04-11 20:42:27 +00:00
Wim Taymans
23d2a4198b Changed to the new props API
Original commit message from CVS:
Changed to the new props API
Other small tuff.
2002-03-30 17:06:26 +00:00
Andy Wingo
3e882a4bf3 filter newlines out of GST_DEBUG statements to reflect new core behavior fixes to adder's caps, again
Original commit message from CVS:
* filter newlines out of GST_DEBUG statements to reflect new core behavior
* fixes to adder's caps, again
2002-03-24 22:07:03 +00:00
Andy Wingo
12bac12e5c added plugin_desc structures to libs, which makes their locations cached in the registry. this speeds plugin loading ...
Original commit message from CVS:
* added plugin_desc structures to libs, which makes their locations cached in the registry. this
speeds plugin loading considerably, especially on uninstalled versions.

* put the lib path before all others, for speed reasons.

* some fixes to adder's caps.

* added linefeeds (\n) to GST_DEBUG strings to match GST_INFO behavior. this is more sane. all
code will need to be converted. i think some perl can do this.
2002-03-24 20:55:44 +00:00
Andy Wingo
598d929eec s/Gnome-Streamer/GStreamer/
Original commit message from CVS:
s/Gnome-Streamer/GStreamer/
2002-03-20 21:45:03 +00:00
Andy Wingo
1049c792c8 removal of //-style comments don't link plugins to core libs -- the versioning is done internally to the plugins with...
Original commit message from CVS:
* removal of //-style comments
* don't link plugins to core libs -- the versioning is done internally to the plugins with the plugin_info struct,
and symbol resolution is lazy, so we can always know if a plugin can be loaded by the plugin_info data. in theory.
2002-03-19 04:10:06 +00:00
Andy Wingo
1ee4a02043 s/@GST_PLUGIN_LDFLAGS@/$(GST_PLUGIN_LDFLAGS)/ @-substitued variables variables are defined as make variables automagi...
Original commit message from CVS:
s/@GST_PLUGIN_LDFLAGS@/$(GST_PLUGIN_LDFLAGS)/

@-substitued variables variables are defined as make variables automagically,
and this gives the user the freedom to say make GST_PLUGIN_LDFLAGS=-myflag
2002-03-19 01:39:43 +00:00
Wrobell
40adafc6f7 - plugins are built without versioning info
Original commit message from CVS:
- plugins are built without versioning info
2002-01-31 00:22:12 +00:00
Thomas Vander Stichele
460f0a863f more distcheck fixes
Original commit message from CVS:
more distcheck fixes
2001-12-25 22:30:51 +00:00
Thomas Vander Stichele
50f9df6c66 more fixes
Original commit message from CVS:
more fixes
2001-12-23 20:21:19 +00:00
Thomas Vander Stichele
2c7cc0f6e8 idct fixes
Original commit message from CVS:
idct fixes
2001-12-23 20:14:18 +00:00
Thomas Vander Stichele
5a1b3887bc more fixes
Original commit message from CVS:
more fixes
2001-12-23 19:40:53 +00:00
Thomas Vander Stichele
0b25ad6ec1 forgot this
Original commit message from CVS:
forgot this
2001-12-23 14:57:56 +00:00
Thomas Vander Stichele
72387bfa1a dvdread added more fixes for libs stuff
Original commit message from CVS:
dvdread added
more fixes for libs stuff
2001-12-23 12:29:33 +00:00
Thomas Vander Stichele
1e418567fa made changes everywhere to accomodate for the headers being in <gst/(lib)/...> we'll need to conclude this fast becau...
Original commit message from CVS:
made changes everywhere to accomodate for the headers being in
<gst/(lib)/...>
we'll need to conclude this fast because we will also need to change stuff in core real soon for the libs in order to fix everything
and I can't do it right now because I disabled all of the plugins here ;)
2001-12-23 12:18:18 +00:00
Thomas Vander Stichele
8f0470580f adding resample lib
Original commit message from CVS:
adding resample lib
2001-12-23 10:44:28 +00:00
Thomas Vander Stichele
1fe98d6244 audio lib removed from this dir
Original commit message from CVS:
audio lib removed from this dir
2001-12-22 23:44:44 +00:00
Thomas Vander Stichele
7ef65f7b85 moving and renaming we put the libs in the source in gst-libs/gst/(dir) the headers get installed in prefix/include/g...
Original commit message from CVS:
moving and renaming
we put the libs in the source in gst-libs/gst/(dir)
the headers get installed in prefix/include/gst/(dir)
the libs are installed in prefix/lib/gst
with a libgst prefix
the sources should be without the gst prefix
as per irc agreement
please comment if this sounds like a bad idea ;)
2001-12-22 23:43:34 +00:00
Wrobell
fd6261f66a - clean up
Original commit message from CVS:
- clean up
2001-12-21 15:52:40 +00:00
Thomas Vander Stichele
e4848fa40a adding Makefile
Original commit message from CVS:
adding Makefile
2001-12-21 12:35:18 +00:00
Thomas Vander Stichele
92f5a8ac6c fixed some GST_LIBS stuff added audiofile added gst-libs/audio building
Original commit message from CVS:
fixed some GST_LIBS stuff
added audiofile
added gst-libs/audio building
2001-12-21 11:46:15 +00:00
Thomas Vander Stichele
1fcdd0a62d added audiofile and audio library
Original commit message from CVS:

added audiofile and audio library
2001-12-21 10:35:10 +00:00
Thomas Vander Stichele
80d42031b2 audio library
Original commit message from CVS:
audio library
2001-12-21 10:12:38 +00:00