Commit graph

2671 commits

Author SHA1 Message Date
Sebastian Dröge
7dbefedeef videorate: Add Since marker for the new skip-to-first property 2009-09-11 07:38:28 +02:00
Olivier Crête
f35c5bc1e7 videorate: Make videorate work with a live source
Add a property that makes videorate skip to the first buffer it
receives instead of padding the stream from segment start to the
first real buffer.

Fixes bug #567928.
2009-09-11 07:36:10 +02:00
Sebastian Dröge
662a31983f playbin(2): Document that the volume property uses a linear scale
Fixes bug #571610.
2009-09-10 16:56:14 +02:00
Benjamin Otte
8939bc6c82 ffmpegcolorspace: Handle YVU9/YUV9 conversion with odd widths
A green border could be visible when converting to Y444 or RGB, because
the last chroma samples weren't copied correctly
2009-09-10 10:56:29 +02:00
Benjamin Otte
7ed2531b27 videotestsrc: Fix YVU9 and YUV9
- Buffer sizes were computed different from ffmpegcolorspace
- Green bar on right size for widths not divisable by 4
2009-09-10 10:45:06 +02:00
Benjamin Otte
35fdfcc625 videotestsrc: Fix image for odd widths in some formats
videotestsrc rounds chroma down. This causes it to omit the last chroma
value completely for odd widths when the chroma is downsampled.
This patch special cases the last pixel to not be rounded down.
2009-09-10 10:45:06 +02:00
Stefan Kost
7a3797f332 docs: tell a biit more about uri-decodebin and buffering 2009-09-10 10:30:23 +03:00
Wim Taymans
2a5cd16eb1 fix whitespace 2009-09-08 13:02:46 +02:00
Tim-Philipp Müller
f051514f69 typefinding: disable typefinder for headerless flac
Disable headerless flac typefinder as long as it happily typefinds anything
including /dev/urandom as flac and as long as it's not particularly useful
given that such streams don't really exist in the wild.

Also fix up some comments so that gtk-doc doesn't complain about them.
2009-09-07 17:13:12 +01:00
Wim Taymans
f1b209f409 typefind: fix midi typefinding
We already have a audio/midi typefinder so don't override it with the midi in
RIFF typefinder or else we fail to detect plain midi files.
2009-09-04 15:48:06 +02:00
Wim Taymans
9e83339cf6 uridecodebin: do buffering for more uris
Add ssh://, ftp://, sftp://, myth:// to the list of uris that require
buffering.

Fixes #594020
2009-09-04 11:33:04 +02:00
Sebastian Dröge
a69ffb5886 typefindfunctions: Add typefinder for Midi inside RIFF
This is a standard Midi file format that should be supported by
all Midi decoders and also has the mimetype audio/mid according to
the Midi specification homepage.

Fixes bug #594094.
2009-09-04 07:36:10 +02:00
Tim-Philipp Müller
4cacc441d8 typefinding: move gio-based xdg mime typefinder from -bad to -base
Its purposes is mainly to avoid false positives (e.g. mp3 typefinder
reporting a 20% probability and somesuch). Won't be registered if
the gio plugin has been disabled via ./configure --disable-gio.
2009-09-03 09:01:47 +01:00
Tim-Philipp Müller
889c318798 subparse: GstAdapter is not a GstObject and should be freed with g_object_unref 2009-09-01 15:16:17 +01:00
Stefan Kost
e7368354d5 adder: improve caps filter functionality. Fixes #590146.
Also use the capsfilter if there is no src-peer as the caps constrain what
we can do. Don't create any_caps as a default, as we check for NULL to skip the
filtering. This is a (small) performance regression as we always intersect
otherwise.
2009-08-31 22:48:01 +03:00
Sebastian Dröge
2194166e05 decodebin2: Post missing plugin messages before any error messages 2009-08-31 11:10:55 +02:00
Marc-André Lureau
605f3c2942 Bug 593035 - set IN_CAPS for streamheader buffer 2009-08-28 08:44:55 +01:00
Sebastian Dröge
460dc94d23 playbin: The internally linked pad of the selector might be NULL in some cases 2009-08-26 16:56:19 +02:00
Sebastian Dröge
67a0ef9b3a playbin: Fix iterate internal linked pads functions for the stream selectors
This now used the new gst_iterator_new_single() function and as a side effect
fixes bug #592864.
2009-08-26 16:45:49 +02:00
Sebastian Dröge
3c8ff21ba2 typefindfunctions: Detect AVF files as RIFF files too
AVF is valid RIFF but has AVF0 as first fourcc instead of RIFF.

Partially fixes bug #593117.
2009-08-26 09:10:19 +02:00
Sebastian Dröge
1e450f21f8 audioresample: Fix drain processing
In case we have to convert internally don't process output length input samples
but history length input samples.
2009-08-26 09:10:18 +02:00
Sebastian Dröge
2e585ac7ac audioresample: On the first buffer we need discont handling
Otherwise we won't get upstream timestamps and everything and all
output buffers would have -1 timestamps.
2009-08-26 09:10:18 +02:00
Руслан Ижбулатов
5d96fd4bf1 subparse: Remove dependency on regex.h as it's not used anyway
Fixes bug #592544.
2009-08-26 09:10:17 +02:00
Kipp Cannon
86b4c51c8c audioresample: Fix buffer overflow when pushing the drain 2009-08-26 09:10:17 +02:00
Kipp Cannon
a69068d70d audioresample: Fix timestamp drift
Fixes bug #591934.
2009-08-26 09:10:17 +02:00
David Schleef
0e9bc5125a Remove Ronald Bultje from Authors field
Replaced with "GStreamer maintainers
<gstreamer-devel@lists.sourceforge.net>" or just removed,
depending on the number of other authors.
2009-08-24 11:37:01 -07:00
Wim Taymans
c3ebeec5a5 playbin2: fix refcounting of _get_sink()
g_value_set_object() increases the refcount of the sink, which is not needed
because the object should already be refcounted. Make sure this is always the
case and use g_value_take_object().

Fixes: #592884
2009-08-24 15:08:36 +02:00
Sebastian Dröge
0c1fa2e8ab streamselector/inputselector: Use iterate internal links instead of deprecated get internal links 2009-08-19 17:23:21 +02:00
Sebastian Dröge
b7fa34a279 playsink: Also send SEEK events directly to a subpicture sink 2009-08-18 11:15:41 +02:00
Sebastian Dröge
b5f84c0637 playsink: If a custom text sink is used, send events to it too
Before, SEEK events would be sent to the video sink, which wouldn't
be linked in any way to the subtitle part of the pipeline and
subparse would never see the SEEK event. This would then seek
the audio/video but the subtitles would continue from the old
position instead.

Fixes bug #591664.
2009-08-18 08:39:02 +02:00
Sebastian Dröge
6aa731cb48 uridecodebin: Make missing plugins emit a warning message, not an error message
The problem with an error message is, that it will stop playback completely
while it could be that only a audio decoder plugin is missing and the video
could be played with the available plugins.

See bug #591677.
2009-08-18 08:20:28 +02:00
Sebastian Dröge
a6b1e0b645 uridecodebin: Post a correct error message for unknown types
Before we had STREAM/WRONG_TYPE but it's really CORE/MISSING_PLUGIN
because a plugin is missing and nothing else is wrong.

Also make it an error instead of a warning.

Really fixes bug #591677.
2009-08-13 17:42:07 +02:00
Sebastian Dröge
504f8dc9c7 uridecodebin: Post a missing plugin message additional to the error message on unknown types
Fixes bug #591677.
2009-08-13 15:55:25 +02:00
Tim-Philipp Müller
4871cd9254 playbin2: fix error message string
Fixes #591577.
2009-08-13 10:59:35 +01:00
Mark Nauwelaerts
188d698449 decodebin2: avoid assertion failure on empty/NULL caps 2009-08-12 13:39:12 +02:00
Sebastian Dröge
3b4c35e319 typefindfunctions: Also detect SVG by the <svg> starting tag
Not all SVG images have the DOCTYPE specified.
2009-08-12 12:11:08 +02:00
Sebastian Dröge
88a55e6dae subparse: Allow . instead of , as millisecond delimiter in srt subtitles
Fixes bug #591207.
2009-08-09 12:13:16 +02:00
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
8cd1b5209b gst: Remove dead assignments and resulting unused variables 2009-08-08 15:54:02 +02:00
Sebastian Dröge
141c3f52cd typefindfunctions: Add typefinders for many game sound console formats supported by gme
These are AY, GBS, GYM, KSS, SAP and VGM. SPC and NSF already had typefinders.
2009-08-07 09:33:34 +02:00
Siarhei Siamashka
720a927f38 ffmpegcolorspace: support for direct conversion from uyvy422 to rgb formats 2009-08-06 15:09:44 +03:00
Stefan Kost
007da06645 playbin2: smarter sink selection. Fixes #588523
Don't do fallbacks if application specified a sink element. When doing the
fallback use configured default elements instead of hardcoded linux only
elements. Improve error messages accordingly.
2009-08-06 15:07:02 +03:00
Mark Nauwelaerts
ff998f24db queue2: post error message when pausing task if so appropriate
If a downstream element returns an error while upstream has already
put all data into queue2 (including EOS), upstream will no longer
chain into queue2, so it is up to queue2 to perform some
EOS handling / message posting in such cases.  See #589991.
2009-08-06 13:39:19 +02:00
Tim-Philipp Müller
85a08d8dc2 typefinding: fix postscript typefinder probability
Two bytes for a rare format hardly warrants MAXIMUM typefinding
probability, POSSIBLE seems more appropriate.
2009-08-06 09:57:25 +01:00
Sebastian Dröge
060f9c07e5 subparse: Implement POSITION query 2009-08-06 06:43:38 +02:00
Sebastian Dröge
7e119e46e5 subparse: Implement SEEKING query 2009-08-06 06:43:38 +02:00
Sebastian Dröge
76571840ef typefindfunctions: Add SVG typefinder 2009-08-06 06:43:35 +02:00
Sebastian Dröge
5c52513aab typefindfunctions: Add postscript typefinder 2009-08-06 06:43:35 +02:00
Sebastian Dröge
37839ee2b3 typefindfunctions: Use static caps again for MPEG4 typefinding 2009-08-06 06:43:35 +02:00
Arnout Vandecappelle
ce24ac4ed0 typefindfunctions: Implement better & more flexible MPEG4 typefinding
This detects more MPEG4 streams as MPEG4.

Fixes bug #556537.
2009-08-06 06:43:35 +02:00
Sebastian Dröge
cac4b032c3 videoscale: Restrict width/height to 2^15 - 1
Otherwise integer overflows will happen, resulting in segmentation faults.

Fixes bug #590243.
2009-08-06 06:43:34 +02:00
Sebastian Dröge
6b63053be1 ffmpegcolorspace: Fix indention of template header 2009-08-06 06:43:34 +02:00
Benjamin Gaignard
2f4c65bb06 typefindfunctions: Fix typefinding of SDP files
Fixes bug #589574.
2009-08-06 06:43:33 +02:00
Kipp Cannon
4689acd68f audioresample: Take the output offsets from the input if possible
Fixes bug #588915.
2009-08-06 06:43:33 +02:00
Sebastian Dröge
b69f5e2c66 videoscale: Make sure to allocate enough memory for the temporary buffer
and fix scaling of odd-height interlaced video.
2009-08-06 06:43:32 +02:00
Sebastian Dröge
c51d2febd3 videoscale: Fix interlaced scaling for I420
...and some other minor mistakes in the previous change.
2009-08-06 06:43:32 +02:00
Sebastian Dröge
164b90f9d0 ffmpegcolorspace: Include interlacing information in the AVPicture
This later allows to handle interlaced AVPicture different than
progressive ones which is needed for horizontally subsampled YUV
formats, see bug #589242.
2009-08-06 06:43:32 +02:00
Sebastian Dröge
33c490f4b9 videoscale: Add support for interlaced content
videoscale is not mixing content of two seperate fields anymore
and does scaling on every field separately.

Fixes bug #588761.
2009-08-06 06:43:31 +02:00
Tim-Philipp Müller
e199d7e1cd typefinding: fix detection of fLaC id packet in broken flac-in-ogg
There are flac-in-ogg files without the usual flac packet framing
and these files just have a 4-byte fLaC ID packet as first packet.
We need to recognise the type just from these four bytes if we
want oggdemux to recognise these streams correctly.
2009-08-01 19:01:39 +01:00
Stefan Kost
7205bbc031 adder: reset pending flush-stop flag in state_changed. (mostly) Fixes #590146. 2009-07-30 13:45:42 +03:00
Tim-Philipp Müller
789f5b0488 playbin: rename GType of stream selector pad to avoid clash with input-selector from -bad
Rename the GType of the pads of playbin's internal stream selector
element so they don't use the same type name as input-selector's
pads. Fixes #589622.
2009-07-27 14:18:27 +01:00
Tim-Philipp Müller
2670f23812 typefind: recognise Kate spu subtitles as well
Recognise spu-subtitles, SUB and K-SPU as valid categories for
Kate subtitles as well.
2009-07-24 09:42:05 +01:00
Wim Taymans
0bb9b75a75 audiotestsrc: call send_event directly
We can't call gst_element_send_event() from a streaming thread as it gets the
state lock. Instead call the send_event method directly until we have a nice API
for this in basesrc.

Fixes #588746
2009-07-20 13:15:32 +02:00
Edward Hervey
3708ca37a8 gstadder: Don't forget to free pending events on flush/dispose.
Fixes #588747
2009-07-20 12:42:32 +02:00
Edward Hervey
196b38d4ef audiotestsrc: Make sure tags are properly serialized. Fixes #588746
We do this by letting the basesrc base class handle the tags.
2009-07-20 08:47:50 +02:00
Edward Hervey
50b0cf2c03 adder: Collect incoming tag events and send them after newsegment. Fixes #588747 2009-07-19 10:49:17 +02:00
Wim Taymans
3886a83f0e queue2: fix leak and improve buffering
Keep track of the max requested position and compare this to the write position
in the temp file to get the current amount of buffered data.
Fix memleak of all incomming buffers.

Fixes #588551
2009-07-15 20:56:10 +02:00
Tim-Philipp Müller
5366b61bfc playbin2: use private copy of input-selector
We shouldn't really depend on elements from -bad for stream
selection in playbin2, so use a private copy of input-selector
until the selector plugin is ready to be moved to -base or -good.
Fixes #586356.
2009-07-15 17:42:46 +01:00
Tim-Philipp Müller
e1df8d0691 playback: add private copy of the input-selector from gst-plugins-bad
Not hooked up yet though. See #586356.
2009-07-15 17:27:28 +01:00
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
31b0c658e5 uridecodebin: treat uri-schemas incasesensitive
Treat uri-schemas incasesensitive. This is mandated in rfc2396 section 3.1.
Fixes not showing buffering messages e.g. for HTTP://...
2009-07-13 21:56:46 +03:00
Stefan Kost
94baad7490 adder: add since tags to docs 2009-07-10 23:27:11 +01:00
Wim Taymans
084357dfb8 queue2: flush differently, avoiding deadlocks
Don't flush the file by closing and opening it but instead use g_freopen. This
avoids a deadlock in shutdown because we emit the temp-location property change
with the wrong lock held.
2009-07-10 21:01:39 +01:00
Wim Taymans
bede11dbc3 uridecodebin: Fix template construction
Fix the construction of the temporary filename construction as the application
name can be NULL and we don't want a separator between the prgname and the
template.
2009-07-10 20:26:22 +01:00
Wim Taymans
4403cf4efb playbin2: add support for progressive download
Add a new playbin2 flag (initially disabled) to enable progressive download
buffering in uridecodebin.
2009-07-10 20:26:22 +01:00
Wim Taymans
f4d78328dd uridecodebin: add download property
Add a download property that will attempt to configure queue2 into progressive
download buffering.
Make sure we only enable download buffering for quicktime and flv formats.
2009-07-10 20:26:22 +01:00
Wim Taymans
df58d6a39e queue2: add temp-template property
Add a new temp-template property so that queue2 can securely allocate a
temporary filename. Deprecate the temp-location property for setting the
location but still use it to notify the allocated temp file.
2009-07-10 20:26:22 +01:00
Stefan Kost
725bd20045 adder: add a caps-property to avoid to need to plug a capsfilter afterwards
Adder can only handle one common format accross the pads. Thus one needed to add
a capsfilter afterwards and manage the caps. Now one can simply set the caps on
the property.
2009-07-10 20:06:28 +01:00
Sebastian Dröge
3d751d190d playbin2: Make it possible for READY->PAUSED to succeed after it failed the first time
If READY->PAUSED failed in the source element we would've swapped
the current and next group already. To allow READY->PAUSED to succeed
after the first failure we have to swap the current and next group
back again. This also ensure that we're again in the same state
as before the failed state change and not at the next group.

This was especially a problem for playbin2 pipelines that use the
new mounting support in giosrc as the source would fail for READY->PAUSED
the first time, the application mounts the location and then tries
to go READY->PAUSED again (and this time it would succeed).

Fixes bug #588078.
2009-07-10 17:08:40 +02:00
Philip Jägenstedt
fa0a5a667f audioconvert: Fix compilation when debugging is disabled
Fixes bug #587980.
2009-07-08 15:08:32 +02:00
Stefan Kost
da27fd57e8 adder: keep sending newsegments after seeking
Adder sends with timestamps from 0 upwards. After seeking we need to send
new-segments to get correct positions-queries.
2009-07-06 22:35:14 +01:00
Edward Hervey
c3adf88621 adder: Call set_flushing(TRUE) for flushing seeks *when* the streaming is stopped.
This ensures that collectpads' cookie is properly updated so that when the streaming
threads will restart and be checking for the flushing status of all pads there will
be no inconsistent state.
2009-07-05 21:32:20 +02:00
Wim Taymans
e8598d24e5 uridecodebin: make fd:// uri use buffering too
fd:// usually operate in push mode only and are thus suitable for buffering.
2009-06-30 18:44:44 +02:00
Stefan Kost
c1f46ea29e volume: include "1.0=100%" in property description 2009-06-30 14:46:38 +03:00
Stefan Kost
aab2e110a1 playsink: remove unused property defs 2009-06-30 14:45:51 +03:00
Jan Schmidt
ee7fd4c28d playsink: Avoid a segfault when the video sink fails to start
Don't attempt to display the subpictures and segfault when the
video sink failed to start (and hence the videochain is NULL).
2009-06-29 14:35:03 +01:00
René Stadler
61441ff183 playbin2: fix initial volume handling also when reusing the element
This is a follow-up to commit 452988, making it work correctly when the audio
chain is reused.
2009-06-27 16:36:11 +03:00
Stefan Kost
f8506f9e1f screenshot: don't leak message 2009-06-26 13:14:52 +03:00
Tim-Philipp Müller
8b94cd3934 typefinding: lower the h264 typefinder's probability
A NEARLY_CERTAIN is absolutely not warranted given the kind
of things it checks for. Even a LIKELY is probably not entirely
appropriate.
2009-06-25 12:09:59 +01:00
Wim Taymans
09737d728b adder: only unflush when we flushed before
Ass suggested by Stefan Kost:
Keep track of when the sinkpad was set to flushing and unflush the pad when an
upstream flushing seek failed.
2009-06-23 18:08:44 +02:00
Tim-Philipp Müller
5974042bba uridecodebin: fix leak when the source fails to change state 2009-06-23 15:12:50 +01:00
Wim Taymans
c933933f09 ssaparse: avoid leaking all buffers 2009-06-23 12:40:56 +02:00
Sebastian Dröge
dc706f7f2f audiotestsrc: Name gaussian noise "gaussian-noise" instead of just "gaussian"
Also make all the function arrays constant.
2009-06-21 19:43:18 +02:00
Kipp Cannon
620391b300 audiotestsrc: Add support for generating gaussian white noise
This patch adds support for stationary white Gaussian noise.
The Box-Muller algorithm is used to generate pairs of independent
normally-distributed random numbers.

Fixes bug #586519.
2009-06-21 12:29:03 +02:00
Jan Schmidt
0ed6ec3902 ffmpegcolorspace: Fix NV12 and NV21 transformations
Fix some stride problems, fix the nv12 to nv21 direct transformation,
and implement a direct conversion to yuv444 to save CPU.
2009-06-20 23:48:22 +01:00
Jan Schmidt
ad947b2436 videotestsrc: Fix NV12 painting for odd strides/heights 2009-06-20 23:48:22 +01:00
Wim Taymans
85dbf93515 adder: more seeking fixes.
When a seek failed upstream, make sure the adder sinkpad is set unflushing again
so that streaming can continue.
We only have a pending segment when we flushed.
Set the flush_stop_pending flag inside the appropriate locks and before we
attempt to perform the upstream seek.
Add some more comments.
Use the right lock to protect the flags in flush_stop.

See #585708
2009-06-17 11:22:51 +02:00
Sebastian Dröge
62f43a1c52 decodebin2: Free iterator after removing all groups 2009-06-17 07:24:53 +02:00
Wim Taymans
c4d729a4da playbin2: set smarter target state on uridecodebin
Set the target state of the newly added uridecodebins to somthing else that
PAUSED so that we keep their state in sync with the playsink state.

Fixes #585268
2009-06-16 18:20:06 +02:00
Wim Taymans
a31c3bfc60 playsink: set the sink flag on the element 2009-06-16 18:20:05 +02:00
Wim Taymans
7a82caebd2 uridecodebin: add debug message 2009-06-16 18:20:05 +02:00
Stefan Kost
fd36634f88 adder: send flush_stop when seeking failed
At least do the fix to sent the flush_stop when seeking failed to ensure we
keep no pads flushing. before it was send when the seeking worked which is just
plain wrong and was not the intention.
2009-06-15 11:45:19 +03:00
Wim Taymans
45084bf579 adder: send flush-stop earlier
When no flush-stop has been sent by upstream, we have to send one ourselves to
continue playback. Do this as soon as the collect function is called instead of
after we possibly pushed segment events (that got then flushed out)
2009-06-12 16:31:00 +02:00
Wim Taymans
6a7d0ebf2a playsink: update for new step API 2009-06-12 13:52:02 +02:00
Tim-Philipp Müller
9ca2bf36de subparse, ogmparse: post tags with GST_TAG_SUBTITLE_CODEC
Make subtitle parsers post a taglist with codec tags, so the application
knows what kind of subtitle a subtitle stream is. Fixes #576552.
2009-06-11 22:32:28 +01:00
Jan Schmidt
79e97ec5ec playbin2/uridecodebin: Fix connection-speed propagation
uridecodebin expects the passed connection-speed value in kbps, so we
need to divide the value stored in bps by 1000. Also, lower the upper
limit on the properties to the value that we can actually store in our
internal guint (which is plenty high enough)
2009-06-10 17:05:18 +01:00
Tim-Philipp Müller
40bea96ff6 subparse: recognise more subrip timestamp variants
Be even less restrictive in what we accept for .srt timestamps when
typefinding and parsing subrip subtitles and add a unit test for
the 'new' format. Fixes #585197.
2009-06-10 14:41:41 +01:00
Wim Taymans
ef1030ee6e decodebin2: make sure varargs are of right type
Explicitly cast the variables to g_object_set to their right types.
2009-06-05 18:13:25 +02:00
Wim Taymans
f444f0edce decodebin2: increase stream probing queues
When we are probing for streams, we want to set the queue size in such a way
that we can scan a maximum amount of data without consuming too much memory.
Therefore, remove the time limit on the queue and only stop scanning after 2MB
of data.

See #584104.
2009-06-05 16:49:58 +02:00
Sebastian Dröge
28b366db00 playbin2: API: Add {audio,video,text}-tags-changed signals
Fixes bug #584686.
2009-06-04 08:59:28 +02:00
Wim Taymans
49ee8a2d05 playbin2: also set custom text and subp sinks
Set the custom subpicture and text sinks along with the custom audio and video
sinks when needed.
Fix a little docs blurb too.
2009-06-03 12:45:08 +02:00
Edward Hervey
31e7bf098d typefindfunctions: Fix caps for ogg typefinder. 2009-06-02 09:54:23 +02:00
Wim Taymans
ea97973efe add framestepping to playbin2 and seek 2009-06-01 11:31:49 +02:00
Sebastian Dröge
a9c405f5fa ffmpegcolorspace: Add a lot more conversions from/to 16 bit grayscale 2009-05-30 14:18:56 +02:00
Jan Schmidt
8900ada0eb playbin2: Have playbin recognise PGS subpicture streams
Recognise PGS subpicture streams and connect them to the SPU pad
in playsink. Unfortunately this fails badly with negotiation errors
if the SPU is not recent enough to support the stream. I'm not sure
how to add format negotiation in yet.
2009-05-29 00:09:15 +01:00
Jan Schmidt
47d7464b10 decodebin/uridecodebin: Recognise subpicture/x-pgs pads and output them. 2009-05-28 22:07:30 +01:00
Tim-Philipp Müller
effa5e69d9 playbin2: fix volume handling for audio sinks without "volume" property
When using an audio sink without a "volume" property, volume control
would only work for the first song. For the next song, we'd try to
re-use the existing audio chain, but inadvertently set chain->volume
to NULL instead of to the existing volume element.
2009-05-28 20:49:22 +01:00
Tim-Philipp Müller
18fc3e6e18 playbin2: cosmetic change to avoid unnecessary line breaks
Looks nicer and works around gst-indent silliness.
2009-05-28 17:05:55 +01:00
Wim Taymans
3f20b0522a playbin2: don't lose the ref to the volume element
Only release the ref to the volume element when it is controled by a sink. For
software volume we never have to fear that it will change.
2009-05-28 17:21:35 +02:00
Tim-Philipp Müller
7620d8800d playbin2: actually use configured audio/video sinks
playbin2 inadvertently used autoaudiosink and autovideosink up to now,
since it would overwrite the sinks configured via the "audio-sink"
and "video-sink" properties with the stream-specific group sinks when
configuring the outputs. Those are usually NULL however, so that would
overwrite the configured sinks with NULL which makes playbin2 then
default to the auto sinks. Fix this by keeping a reference to each
configured sink in playbin2 and setting up the right sinks depending
on whether there is a stream-specific sink or not.

Fixes #584020.
2009-05-28 15:30:27 +01:00
Wim Taymans
650215bcfd decodebin2: remove leftover elements
Remove all of the elements inside decodebin2 when goint to READY and NULL.
Makes decodebin2 reusable.
Fixes #583750
2009-05-27 18:12:10 +02:00
Wim Taymans
de06a6adb5 playbin2; release refs to volume/mute properties
Release the refs to the volume and mute property elemens before setting the
child elements to READY or NULL.
Fixes #583318
2009-05-27 15:45:25 +02:00
Wim Taymans
4fad939053 gdppay: set caps on outgoing buffers
Set caps on outgoing buffers because NULL caps confuse basetransform.
Fixes #583867
2009-05-27 12:10:05 +02:00
Sebastian Dröge
2e5c946501 videoscale: Add support for 16 bit grayscale in native endianness 2009-05-26 15:26:30 +02:00
Sebastian Dröge
9d02d1d8ab ffmpegcolorspace: Add support for 16 bit grayscale in little/big endian 2009-05-26 15:26:30 +02:00
Sebastian Dröge
d48a5614f4 videotestsrc: Add support for 16 bit grayscale in native endianness 2009-05-26 15:26:29 +02:00
David Schleef
1dae15d762 Run liboil benchmark multiple times
The statistics function requires multiple runs, otherwise
it causes a divide by zero error.
2009-05-22 17:34:56 -07:00
Stefan Kost
45298860fc playbin2: fix initial volume and mute handling
Use two flags to remember volume/mute changes at times when we don't have the
audiochain yet (e.g. construction). Only set values when they were actualy
changed. This makes pulseaudio's stream restore functional.
2009-05-22 15:49:14 +03:00
Wim Taymans
5181760120 tcpclientsrc: this is not a live source
Don't mark us as a live source because we are not.
2009-05-19 18:43:45 +02:00
Stefan Kost
4228ba0c6b adder: only send flush_stop when seek failed
This is still not the ultimate fix. Added some comment to explain the troubles.
2009-05-19 18:44:01 +03:00
Stefan Kost
ef56ebad48 adder: send flush_stop to match flush_start
Adder was relying that something else sends a flush stop. When using adder with
a livesource it was not getting a flush_stop and thus all pads downstream where
keept flushing. Mark a pending flush_stop and send it when we are working on
the new segment back in the streaming thread.
2009-05-19 16:49:35 +03:00
Stefan Kost
eeb3300383 decodebin: use iterators instead of list
The list api is deprecated. Use threadsafe iterators instead.
2009-05-19 16:49:35 +03:00
Wim Taymans
7d049bc29f uridecodebin: configure caps on decodebin2
Implement the caps property by setting the configured caps on new decodebin2
objects.

Fixes #582749
2009-05-19 15:35:54 +02:00
Wim Taymans
e685a9e86d decodebin2: avoid some _caps_ref in some cases
Only mess with the caps refcount when we configure different caps.
2009-05-19 15:34:38 +02:00
Wim Taymans
8f2f705c21 uridecodebin: fix potential caps leak
Free the user-configured caps in finalize.
2009-05-19 15:27:12 +02:00
Wim Taymans
f11edb626f uridecodebin: add queue after cdda://
Add a queue2 after the raw output pads of certain sources such as those for uris
like cdda://

No tuning of the queue is done yet as the defaults seem to work fine for me.

Fixes #582528
2009-05-19 15:20:27 +02:00
Edward Hervey
65c046b1ea audioresample: Don't drain remaining buffers after a flush.
If we were resetted (due to a flush), we can not drain the remaining
buffers since they would be pushed before a valid new newsegment event.
2009-05-19 11:20:19 +02:00
Stefan Kost
c94f952056 adder: add more logging and return value checking 2009-05-19 01:13:34 +03:00
Stefan Kost
705b01aa93 adder: handle the return value from iterator_fold 2009-05-19 01:11:45 +03:00
Stefan Kost
591fbbcea7 adder: use the pad in logging as objects
Helps to differenciate between source and sinks pads.
2009-05-19 01:03:44 +03:00
David Schleef
6f686c9ba0 videotestsrc: Add support for v210 and v216 formats 2009-05-15 18:44:37 -07:00
Arnout Vandecappelle
785f748810 multifdsink: add num-fds property
multifdsink::num-fds
2009-05-15 23:07:01 +02:00
Jan Schmidt
768cadf093 playbin2: Fix cdda:// playback
Don't send async-start when the playsink has already been configured
before changing state.
2009-05-14 22:50:53 +01:00
Tim-Philipp Müller
8d326479a5 audiotestsrc: fix broken enum nick - it should have a hyphen
The enum nick should be 'sine-table', not 'sine table'. Technically this is
an API/ABI change I guess, but anyone who was using this and didn't report
it deserves this.
2009-05-12 17:18:37 +01:00
Tim-Philipp Müller
21228a6934 audiotestsrc: seek to the requested byte offset, not the expected byte offset 2009-05-12 15:32:02 +01:00
Tim-Philipp Müller
72c5884f4a audiotestsrc: support more than just one channel 2009-05-12 15:32:02 +01:00
Wim Taymans
956c9f32a3 videorate: handle invalid timestamps better
Handle buffers with -1 timestamps better by keeping track of the en time of the
previous buffer and assuming the -1 timestamp buffer goes right after the
previous one.

when we have two buffers that are equally good, output the oldest buffer once to
minimize latency.

don't try to calculate latency when the input framerate is unknown.
2009-05-12 10:47:17 +02:00
Wim Taymans
a0525fadb4 decodebin2: make subpictures a raw output format
Subpictures are a raw format, we want those pads exposed so that playbin2 can do
the subpicture mixing.
2009-05-12 10:40:11 +02:00
Wim Taymans
e1aa348246 playbin2: make fallback identity silent
Set the signal-handoffs to FALSE and silent to TRUE for the fallback identity
element so that it consumes less CPU.
2009-05-12 10:38:15 +02:00
Wim Taymans
c6f6282fde playbin2: handle custom audiosinks differently
Keep track of the autoplugged custom sinks and configure them in the playsink
element when we have collected all streams.
Also make sure that we only select one custom sink.
When unreffing the internal sink, we don't need to change the state to NULL.
2009-05-12 10:37:45 +02:00
Wim Taymans
0b372dd371 playbin2: unify custom sink get/set functions
Use one function to set/get all of the different sink types.
cleanup up the subpicture chain too.
Allow setting a custom subpicture sink.
2009-05-12 10:36:25 +02:00
Arnout Vandecappelle
192a34af40 typefindfunctions: made mp3_type_find less aggressive
mp3_type_find could suggest already when only a single valid header
was found, if it ran out of data before the end of the next frame.
Therefore, ignore the last found frame if it was incomplete.
Fixes bug #579692.
2009-05-12 09:03:22 +02:00
Wim Taymans
691a52975e playbin2: fix resume after pause
Don't ignore the state change of the children, they might be doing an ASYNC
state change.
2009-05-08 13:11:01 +02:00
Wim Taymans
7c61249ceb multifdsink: fix signature of the add-full signal
The second parameter is a GstSyncMethod enum, not a boolean.
2009-05-07 23:14:56 +02:00
Wim Taymans
52da312fa5 playsink: initialize variable too 2009-05-07 15:19:05 +02:00
Wim Taymans
c05541c195 playbin2: make playsink go ASYNC to PAUSED
Make playsink go async to the PAUSED state instead of relying on uridecodebin
for async behaviour in playbin. This solves some problems (mainly with DVD)
where the pipeline would go to PLAYING before preroll completed, failing to
select the audiosink clock.
Fixes #581727
2009-05-07 14:28:30 +02:00
Jan Schmidt
02a7b31f0e audioresample: Fix buffer size transformations
When calculating the input/output buffer sizes in the transform_size function,
take the number of channels into account, so we don't end up calculating
a buffer size that only contains a partial number of audio frames.

Also, when going from output size to input size, round down rather than
up, so as to calculate the minimum number of samples that *might* yield
a buffer of the intended destination size.

Fixes: #580470 and #580952
2009-05-01 16:47:53 +01:00
Wim Taymans
915b3d139d videorate: clear discont on duplicated buffers
When videorate duplicates a buffer with a DISCONT flag, it copies the discont on
the first pushed buffer but fails to clear it for subsequent buffers. This
causes theoraenc!oggmux and possibly other elements to consider this a discont
stream.

Fix videorate to produce discont as the first buffer and after a flushing seek.

Fixes #580271.
2009-04-28 16:45:07 +02:00
Edward Hervey
71a372c847 decodebin2: Don't reduce max-size-time of exposed groups. Fixes #579912
The 2s limit is way too small for a lot of files (which have an interleave
in time of between 3 and 5s). Instead, leave it to the initial 5s value
and reduce the other limits (allowing us to stay memory-efficient).
2009-04-23 11:54:55 +02:00
René Stadler
22a69b49a3 audioresample: Fix unused variable in compilation with --disable-gst-debug
Fixes: #579668
2009-04-21 22:18:02 +01:00
Wim Taymans
8f8b638d31 playbin: only use raw_decoding_mode when it's true
First check the pad caps if they are raw before setting the raw_decoding_mode to
TRUE. Fixes playback of transport streams and other streams that require large
queues.
Fixes #579734
2009-04-21 20:57:34 +02:00
Jan Urbanski
63108730a5 multifdsink: add property to resend streamheaders
Adds a new property in multifdsink, resend-streamheader.

If this property is false, the multifdsink will not send the streamheader if
there's already one set for a particular client.

There are some formats in which every stream needs to start with a certain
blob, but you can't inject this blob at leisure. If the producer wants to
change the blob in question and sets in as the streamheader on the outgoing
buffers' caps, new clients of multifdsink will get the new streamheader, but
old clients will break, because they'll see the blob in the middle of the
stream.

The property is true by default, so existing code will not see any difference.

Fixes #578118.
2009-04-14 17:04:06 +02:00
Wim Taymans
19d30b90d4 multifdsink: add property to handle client write
Add a property to disable listening to client writes. This property is usefull
when other code will deal with reading from the client socket.

API: GstMultiFdSink::handle-read property
2009-04-14 16:53:33 +02:00
Wim Taymans
5eed96dc06 playbin2: clear the target
Clear the target of our ghostpads before we remove the pad from the element.
This to make sure that the internal pad is not left linked to whatever pad we
were ghosted to. This should only be a problem when we leak the ghostpads.

Also release our subpicture pads.

Fixes #577288.
2009-04-14 13:55:52 +02:00
Wim Taymans
4265511b70 decodebin2: fix up the debugs and warnings
Use _OBJECT variants because we can. Go over some log statements and put them in
the right category.
Fixes #567740.
2009-04-14 11:34:49 +02:00
Luca Ognibene
1beabc48e8 multifdsink: fix error in sync-method
Multifdsink did not handle sync-method=latest-keyframe correctly when the
soft-limit is set to -1 (unlimited).
Fixes #578583.
2009-04-12 22:26:33 +02:00
Wim Taymans
1a557e60ea playbin2: fix refcounting of visualisations
See #577794.
2009-04-10 13:42:56 +02:00
Wim Taymans
33ef15fad6 playsink: fix refcounting of custom elements
Sink the custom sinks, let other elements we create be sunken by the bin we add
them to.
Fixes #577794.
2009-04-10 13:27:41 +02:00
Wim Taymans
f25e4e4743 playbin2: handle missing input-selector
Gracefully degrade and disable stream selection when input-selector is
missing.
2009-04-10 12:26:16 +02:00
Stefan Kost
509256dce5 playbin2: better error message on sink failure
If we could create the sinks, but the don't work, don't send the missing plugin
message and report that the state-changed failed.
2009-04-08 16:43:27 +03:00
Michael Smith
a0959afb01 playbin2: don't leak selector when getting current stream numbers. 2009-04-03 10:51:42 -07:00
Jan Schmidt
c2a56e3115 playbin: Add simple 'raw decoding mode'.
Raw decoding mode removes almost all buffering in video and audio queues
when a source providing already decoded video/audio is detected, on the
possibly bogus assumption that such a source should provide sufficient
internal queueing. Fixes playback on some DVDs, and improves it
on all.
2009-04-02 12:18:08 +01:00
Sebastian Dröge
22f99da35c videoscale: Don't read over line ends when taking the last Cr or Cb 2009-04-02 11:10:12 +02:00
Sebastian Dröge
db1236505e videoscale: Don't write to few pixels and don't mix Cr and Cb
Fixes bug #577054.
2009-04-02 10:53:10 +02:00
Tim-Philipp Müller
d271c8de53 audioresample: fix negotiation so that upstream can actually fixate to downstream's rate
If one side has a preference for a particular sample rate or set of sample rates, we
should honour this in the caps we advertise and transform to and from, so that elements
actually know about the other side's sample rate preference and can negotiate to it
if supported. Also add unit test for this.
2009-04-01 15:36:38 +01:00
Tim-Philipp Müller
ce51cad163 docs: add a blurb about redirect messages to playbin2 docs 2009-04-01 15:36:37 +01:00
Sebastian Dröge
ab7d52c053 videoscale: Fix linear scaling for one byte components
Fixes bug #577054.
2009-03-29 12:01:33 +02:00
Sebastian Dröge
954b1713e1 videoscale: Fix 4tap scaling of YUYV and friends 2009-03-29 11:53:40 +02:00
Sebastian Dröge
9d586e0475 videoscale: Rewrite YUYV (and friends) scaling and don't read/write over line ends
Partially fixes bug #577054, there's just one issue left now.
2009-03-28 16:08:44 +01:00
Sebastian Dröge
b5793ccd0d videoscale: Use bilinear instead of 4tap scaling for heights < 4
Partially fixes bug #577054.
2009-03-28 11:57:35 +01:00
Sebastian Dröge
8eb3572eef videoscale: Don't read after the end of a line when lineary scaling YUYV/UYVY/RGB/RGBA
This case is for upscaling a frame with width=1
Partially fixes bug #577054.
2009-03-28 11:57:19 +01:00
Sebastian Dröge
b69231ad21 videoscale: Don't read after the end of a line when lineary scaling YUYV/UYVY
Partially fixes bug #577054.
2009-03-28 11:27:56 +01:00
Sebastian Dröge
1c36c03001 videotestsrc: Initialize buffer memory with zeroes
This prevents valgrind warnings when accessing the "x" parts
of xRGB and friends in other elements that handle (and can handle)
xRGB like ARGB (for example videoscale).
2009-03-28 10:40:43 +01:00
Sebastian Dröge
e2f823c2f3 videocale: Add support for video/x-raw-gray with bpp=depth=8 2009-03-28 10:06:24 +01:00
Sebastian Dröge
dea48dc885 videotestsrc: Add support for generating video/x-raw-gray with bpp=depth=8 2009-03-28 10:01:00 +01:00
Sebastian Dröge
77bc4ff763 ffmpegcolorspace: video/x-raw-gray is the same as the YUV Y800 format 2009-03-28 09:43:23 +01:00
Sebastian Dröge
96b9ec4da0 videoscale: Take the next luma value instead of every second next when scaling UYVY and friends 2009-03-27 19:12:49 +01:00
Sebastian Dröge
12bbcc4db9 videoscale: Add support for v308 YUV colorspace 2009-03-27 19:09:47 +01:00
Sebastian Dröge
92eb02c71f videoscale: Add my copyright to the 4tap scalers 2009-03-27 13:15:11 +01:00
Sebastian Dröge
014f6f1508 videoscale: Enable 4-tap scaling for all supported formats 2009-03-27 13:14:17 +01:00
Sebastian Dröge
77795ac240 videoscale: Implement 4-tap scaling for RGB565 and RGB555 2009-03-27 13:14:00 +01:00
Sebastian Dröge
c1653fc5b0 videoscale: Implement 4-tap scaling for UYVY 2009-03-27 10:47:39 +01:00
Sebastian Dröge
435e9f8b38 videoscale: Implement 4-tap scaling for YUY2 and YVYU 2009-03-27 09:33:58 +01:00
Sebastian Dröge
2045a89350 videoscale: Implement 4-tap scaling for RGB and BGR 2009-03-26 22:14:53 +01:00
Sebastian Dröge
c2d31eb2d1 videoscale: Implement 4-tap scaling for RGBA and other 4 byte formats 2009-03-26 22:08:26 +01:00
Wim Taymans
5ec2d48f0a decodebin2: do some more cleanup
Free the groups when we go to READY.
Allow for NO_PREROLL elements.
2009-03-25 19:01:45 +01:00
Wim Taymans
9bf8277d13 playbin2: add more support for subpictures 2009-03-24 18:29:28 +01:00
Wim Taymans
e7b382c6a9 playbin2: first support for subpictures
Add beginnings of subpicture support.
2009-03-24 17:12:53 +01:00
Wim Taymans
786b0c248c playbin2: blacklist subpictures for now
Blacklist the subpictures until we add support for them.
Add some small debug info.
See #576408.
2009-03-24 12:23:24 +01:00
Wim Taymans
852ace9cce uridecodebin: expose more media types
Expose more media types from a raw source, such as the subpicture and various
text pads.
Small cleanups  and add some more debugging.
See #576408.
2009-03-24 12:19:30 +01:00
Wim Taymans
7cf4e3eb15 playbin2: rescan audio sinks for volume/mute
Rescan the audio sinks for the mute and volume properties.
fixes #576180.
2009-03-24 10:42:04 +01:00
Wim Taymans
14be3f41e2 playbin2: fix reuse of the video chains
When reusing playbin with visualisations, reset the async property on the video
sink because some sinks might dynamically recreate their sinks.
Fixes #576188
2009-03-23 19:40:18 +01:00
Wim Taymans
7628319688 playbin2: allow dynamic swtiching of subtitles
When we have the textpad configured, enable and disable the subtitles by setting
the silent flag on the overlay element instead of trying to remove elements.
See #576187
2009-03-23 17:38:46 +01:00
Wim Taymans
d8003bea06 playbin2: fix dynamic switching of visualisations
Fix the switching of visualisations by requesting and releasing the tee request
pads on demand.
See #576187.
2009-03-23 16:06:43 +01:00
Stefan Kost
3d20bad4ba docs: add examples for tcp elements, also use correct section name. Fixes #564139
Updated the examples in the README to actually work. Add them to api docs. Tests
the api-docs and fix the section names to make the docs actualy show up.
The example for "tcpserversrc" needs review (might be an element bug).
2009-03-23 17:03:38 +02:00
Stefan Kost
46c18b2aa3 indent: fix damange that gst-indent did some time ago 2009-03-23 17:03:37 +02:00
Wim Taymans
7cc84f9205 playbin2: fix linking order
Link after doing the state change and unlink before shutting down. Makes the
window for causing races in toggling the visualisations smaller.
See #576187.
2009-03-23 15:27:27 +01:00
Wim Taymans
779d6f886d uridecodebin: reset counter
reset the number of pending dynamic operations back to 0 when we reuse
uridecodebin.
Fixes #576190
2009-03-23 12:28:18 +01:00
Wim Taymans
554b3aafe4 decodebin2: a pad starts out being not drained.
Mark a new pad as not drained until we get EOS on it.
2009-03-20 15:47:47 +01:00
LRN
23e603f054 win32: fix seeking in large files
Fix Seeking in large files by using the 64-bit seek functions.
Fixes #576019
2009-03-20 14:17:19 +01:00
Wim Taymans
600a810236 decodebin2: recover from failing to add a pad
When we cannot add a pad to the decodebin2 for some reason, print a warning but
continue adding the remaining pads.
2009-03-19 20:31:01 +01:00
Wim Taymans
20468a22c9 decodebin2: more cleanups and docs.
Add some more comments and use g_list_prepend().
2009-03-19 19:35:15 +01:00
Wim Taymans
17e7948325 decodebin2: refactoring and race fixes
Refactor some code so that we can take the right locks and in the right order.
Fixes quite a bit of races already.
2009-03-19 19:19:38 +01:00
Wim Taymans
2f39597919 playbin2: remove the group cond + cleanups
Remove the group GCond that we used for waiting for groups to finish because we
use pad blocking on the selectors and counters instead for waiting for the
groups to complete.

remove the obsolete about_to_finish variable set while emiting the
about-to-finish signal and fix some old comments.

We don't need to take the playbin lock when querying the uridecodebin.
2009-03-19 19:03:25 +01:00
Wim Taymans
563db0fdca decodebin2: add extra dynamic ref for demuxers
When we make a group connected to a demuxer, keep an extra dynamic refcount for
the group which is only decremented when no_more_pads or a multiqueue overrun is
detected. This way we avoid a race between exposing the group while more dynamic
refs are added from new pads.

Fixes #575588.
2009-03-17 19:09:02 +01:00
Wim Taymans
da6a544bdf playbin2: sync state of the sink correctly
Sync the state of the newly added chains to the state of the parent sink element
to avoid lost async-start messages. Fixes cdda:// async-done message storm.
2009-03-17 15:39:23 +01:00
Wim Taymans
a8579ffea0 playbin2: return NOT_LINKED for unselected streams
When streams are not selected in the selector, return NOT_LINKED so that
upstream elements can skip decoding. Only do this for audio and video pads
because for text streams the overhead is smaller and they could come from
external files.
2009-03-17 11:54:40 +01:00
Wim Taymans
5021c930ba playbin: set custom text sink properties
Set the custom sink async=FALSE to not make it participate in preroll because we
are dealing with sparse streams.
Try to set sync=TRUE on the custom text sink.
2009-03-17 11:51:58 +01:00
Wim Taymans
48f7f6b7c2 playbin2: don't try to set invalid stream numbers
Fix a problem with setting the stream numbers because we check for the wrong
range.
See #575239.
2009-03-16 16:42:18 +01:00
Wim Taymans
ba6d3b5aca playbin2: release the shutdown lock
Release the shutdown lock when we wait for other groups to complete or else we
have a deadlock when the other group completes and tries to grab the shutdown
lock.
Fixes #575550.
2009-03-16 16:16:30 +01:00
Tim-Philipp Müller
04a860c6f7 typefinding: make flac typefinder return lower probability for frame headers
The flac frame header typefinder overstates the likelihood of a match, leading
to false positives with e.g. aac streams and PDF files. Reduce probabilty
returned from LIKELY to POSSIBLE for the frame header matchin code.
Fixes #574939.
2009-03-15 19:57:36 +00:00
Tim-Philipp Müller
a622ff74b0 typefinding: improve image/bmp typefinder
Detect more variations and also bail out in more cases where the values
don't make sense. Furthermore, add width/height and bpp to the caps,
because we can.
2009-03-15 19:52:46 +00:00
Wim Taymans
777f8ab1bf playbin2: fix raw elements like cdda://
Fix a fixme with a one liner and make cd playback work again.
2009-03-12 18:27:25 +01:00
Wim Taymans
7849db42b8 playbin2: improve subtitle handling
Add property to playbin2 to configure a custom sink that receives the raw
subtitle buffers instead of using a textoverlay.
Improve the property finding code to make it more usable.
Use property find code to find async properties in custom sinks that are bins.
Improve text overlay code to gracefully handle missing elements.
2009-03-12 17:51:39 +01:00
Tim-Philipp Müller
4cbe4d2c72 typefinding: flac typefinder fixes
Use scan context for initial peek as well. Peek 6 bytes in the initial
peek rather than 5 bytes, to match the length of the memcmp we're doing
on that data later. Return immediately when we found caps from looking
at the beginning of the data - no point in continuing to scan the next
64kB for something matching a frame header.
2009-03-11 13:33:33 +00:00
Stefan Kost
e633c46e95 adder: log details in getcaps like in setcaps 2009-03-10 21:14:43 +02:00
Jonathan Matthew
db7ecda64f typefind: add photoshop typefind functions
Add photoshop typefind functions.
Fixes #574516.
2009-03-09 16:19:40 +01:00
Wim Taymans
72533ecccc decodebin2: only remove pads that were added
Flag pads that were added so that we can see if we need to remove them later or
not.
2009-03-09 15:46:21 +01:00
Mark Nauwelaerts
b7ea2a9105 Unblock blocked ghostpads when shutting down. Fixes #574293. 2009-03-09 13:32:21 +01:00
Edward Hervey
9acf7de5a4 typefind: Use the proper data pointer instead of poking random memory. 2009-03-09 09:08:00 +01:00
Michael Smith
e9e9d82fbe decodebin2: don't stay connected to notify::caps after negotiation
Disconnect the notify::caps signal in our callback (it'll be re-added
if we're not, in fact, finished getting complete caps). Ensures that
caps changes mid-stream (e.g. from an mp3 that changes from
stereo->mono mid-file) don't cause us to try to add a new pad.
2009-03-05 15:44:17 -08:00
Stefan Kost
79771eaba7 adder: add variants for unsigned to fix warnings for unneeded check
For unsigned int out+in can't be < 0.
2009-03-05 12:27:16 +02:00
Stefan Kost
2723c7e4f3 subparse: use the right variable in debug log, encoding is not yet initialized 2009-03-05 10:58:12 +02:00
Stefan Kost
388fa77c11 audioresample: add missing break in event handling, remove dead code 2009-03-05 10:39:33 +02:00
LRN
db596d27a2 subparse: Convert regex code to GRegex code
Fixes: #572993.  Patch author prefers to use an alias, contact
ds if you actually need a real name.

Signed-off-by: David Schleef <ds@schleef.org>
2009-03-02 11:47:39 -08:00
Stefan Kost
46833b9bc7 subparse: don't leak line, if flushing 2009-02-26 18:01:05 +02:00
Stefan Kost
60847e48d7 ffmpegcolorspace: remove unused code/variables 2009-02-26 18:01:04 +02:00
Stefan Kost
5e6447c0ac docs: fix random text after since: tag. Also fix class name to make the docs actual appear. 2009-02-26 10:10:00 +02:00
Stefan Kost
a6ea8280a2 docs: playbin2 has no stream-info 2009-02-26 10:09:59 +02:00
Peter Kjellerstedt
a038a8d46d rtsp, multifdsink: Unify the use of union gst_sockaddr. 2009-02-25 15:45:50 +01:00
Wim Taymans
f5a3387bdb playbin: use flushing pads instead of fakesink
Use the flushing pads on playsink to terminate on shutdown instead of plugging
fakesinks. this should be a little cheaper.
2009-02-25 12:48:53 +01:00
Wim Taymans
747841e97c playsink: Add FLUSHING pad type
Make it possible to request a flushing pad from the playsink. We can eventually
use these flushing pads to quickly terminate the dataflow when we are shutting
down.
2009-02-25 12:48:53 +01:00
Wim Taymans
dbfc80cd6c Release the group lock when setting states
Release the group lock while we perform the state changes on the uridecodebins
because that might trigger callbacks that we need to handle with the group lock
taken. Avoids a possible deadly embrace in some id3/flac files.
Fixes #567396.
2009-02-25 10:08:29 +01:00
Wim Taymans
0b2238b70b Combine finding and creating groups
Combine the search for the current group and optionally creating one into one
function so that we can avoid taking the lock multiple times.
2009-02-25 10:05:38 +01:00
Edward Hervey
2968cc8710 Playbin2: Don't leave unused parameters in debug statements.
Fixes build on macosx
2009-02-25 08:22:00 +01:00
Wim Taymans
b725e1d2c6 Add some G_UNLIKELY because we can
Add a G_UNLIKELY when checking the shutdown variable.
2009-02-24 18:44:54 +01:00
Jan Schmidt
fff6909c1b multifdsink: Fix strict aliasing error using a union 2009-02-24 17:03:08 +00:00
Sebastian Dröge
77a56d5975 ffmpegcolorspace: Add conversion from/to YVYU colorspace
Fixes bug #572872.
2009-02-24 14:06:38 +01:00
Jonas Danielsson
0842dd1c6f ffmpegcolorspace: Add direct UYVY->GRAY8 conversion
The conversion from UYVY to RGB24 and then to GRAY8
is quite slow. Fixes bug #569655.
2009-02-24 13:42:01 +01:00
Mark Nauwelaerts
d24e75f9fa playbin2: fix deadlock when shutting down. Fixes #572577. 2009-02-24 13:30:07 +01:00
Mark Nauwelaerts
bbd66c6baf playbin2/playsink: Set audiotee to PAUSED state in all cases. Fixes #565105. 2009-02-24 10:46:35 +01:00
Sebastian Dröge
6c28744f76 audioresample: Add locking to protect the resampling context
When setting the quality/filter-length while PLAYING the
resampling context will be destroyed and created again in
some cases, which will cause crashes in the transform function
if it's called at that time.
2009-02-15 07:30:17 +01:00
Sebastian Dröge
c080bfae6d ffmpegcolorspace/videotestsrc: Use v308 instead of V308 2009-02-13 10:10:25 +01:00
Sebastian Dröge
65c322edf2 ffmpegcolorspace: Add support for packed 4:4:4 YUV (format=V308)
Only conversions from/to are implemented, which
gives (indirect) support for all possible conversions.

Partially fixes bug #571147.
2009-02-12 19:09:40 +01:00
Sebastian Dröge
79d0fff231 videotestsrc: Add support for packed 4:4:4 YUV (format=V308)
Partially fixes bug #571147.
2009-02-12 19:09:40 +01:00
Michael Smith
4713bb3abc Revert "Remove pad-removed handlers after setting the decodebins to NULL."
This reverts commit b36d8f3e11.

This brought back some deadlocks. A small leak is better, for now. Need to
figure out a way to fix the leak properly.
2009-02-10 20:38:58 -08:00
Michael Smith
41314315c7 playbin2: Fix segfault on notify after group change.
If our group has been switched, then we get a selector active-pad
notification, we don't need to notify.
2009-02-10 17:20:12 -08:00
Michael Smith
a264efc627 playbin2: Look for volume/mute properties recursively in audio element.
Rather than only checking for volume property on the audio sink
directly, recursively look for it on sinks within it (if it's a bin).
Allows use of sink-as-volume-control where the application has supplied
an audio-sink bin that includes a real audio sink internally.
2009-02-10 17:20:12 -08:00
Sebastian Dröge
5fc20b9ec5 videotestsrc: Add support for Y444 (planar 4:4:4 YUV)
Partially fixes bug #571147.
2009-02-10 17:45:59 +01:00
Stefan Kost
f010a38b0d playbin2: implement GST_PLAY_FLAG_NATIVE_{AUDIO,VIDEO}
The flags where present but actually not been taken into account.
2009-02-04 13:56:14 +02:00
Stefan Kost
c6ab453eed audioresample: Add a proper deprecation comment and also drop G_PARAM_CONSTRUCT.
The comment will ensure that is is marked properly in the docs and the
GParamSpecflag was causing a duplicated initialisation of the same value.
2009-02-04 13:56:13 +02:00
Stefan Kost
b08c0a9003 audioresample: Only pull in liboil if its actualy used.
Liboil still has quite significant startup overhead especialy on embedded
platforms. In audioresample it was only used for the profiling timer.
2009-02-04 10:31:21 +02:00
Stefan Kost
080493ccff typefind: Make the flac check more tight to not mistace some aac files for flac. Fixes #570356.
Add comments about the flac format. Tighten the check to not allow values that
refer to headers.
2009-02-03 15:28:50 +02:00
Stefan Kost
0ea2afee42 Allow to configure the resampler function for integer to skip the benchmarking. Fix releasing the intger resampler in benchmark. 2009-02-02 15:45:44 +02:00
Wim Taymans
9996aab207 Fix documentation for autoplug-select
fix the documentation strings for the autoplug-select signal.
Fixes #570142.
2009-02-02 12:54:31 +01:00
Michael Smith
b36d8f3e11 Remove pad-removed handlers after setting the decodebins to NULL.
They do needed cleanup; without this we leak selector requestpads.
2009-01-30 18:30:10 -08:00
Michael Smith
61e81ada2c Unref selector request pad even if we no longer have a selector.
During destruction, we won't have a selector any more, but we still need
to unref the pad to avoid leaking it.
2009-01-30 18:30:10 -08:00
Michael Smith
c799f3f77f Unref source in playbin2's finalize method 2009-01-30 18:30:10 -08:00
Michael Smith
b6cbe7e331 Fix more leaks of pads and elements in gstplaysink.
Don't keep extra references to volume and mute elements; we don't need
to do so.
Ensure we unref pads that we have references to, and release request
pads.
2009-01-30 18:30:10 -08:00
Michael Smith
c34f444174 Avoid leaking all playsinks. Fix some internal leaks.
Playsink was holding references to itself. Don't do that, it's not cool.
Also, free all chains in dispose.
2009-01-30 18:30:10 -08:00
Michael Smith
906502b9bb Unref peer request pad after releasing it, since we hold a reference. 2009-01-30 18:30:10 -08:00
Michael Smith
af8d3c51f0 Fix caps leak in playbin2. 2009-01-30 18:30:10 -08:00
Michael Smith
ef1fa84575 Unref active pad from selector when finding active stream. 2009-01-30 18:30:10 -08:00
Michael Smith
f7abf8ed94 Free uris when finalizing playbin2 instance. 2009-01-30 18:30:10 -08:00
Michael Smith
a2b0229058 Unref pads when iterating over them in analyse_source.
Fixes leak of source's srcpad when using uridecodebin.
2009-01-30 18:30:09 -08:00
Jan Schmidt
5337bc03be Fix compilation warning on Forte 2009-01-30 17:58:15 +00:00
Jan Schmidt
6b1e08f277 Don't do void pointer arithmetic. 2009-01-30 17:16:39 +00:00
Michael Smith
81cc88326f Ensure we have sufficient data when using data scan contexts.
Fixes crashes typefinding things that look like they might contain AAC
data (but probably aren't actually AAC).
2009-01-26 18:02:00 -08:00
Sebastian Dröge
5dfcb63252 Rename files and types from speexresample to audioresample
Rename files and types from speexresample to audioresample
to finish the move and to prevent any confusion.
2009-01-23 12:33:41 +01:00
Wim Taymans
cc8b9ae5e8 Add typefind function for gsm
Because core now supports typefindfactories without a typefind function we can
register a factory fo GSM that will --if all else fails-- assume the file is a
GSM file based on the registered extension.
Fixes #566661.
2009-01-23 11:40:26 +01:00
Wim Taymans
8dd3c2d543 Use more performant link function
We can use gst_element_link_pads() instead of the more generic
gst_element_link() function because we know the pads. This saves some cycles
because the more generic function needs to search for possible compatible caps
etc.
2009-01-23 11:37:45 +01:00
Benjamin Gaignard
336e1346e4 Add typefinder for Mobile XMF. Fixes bug #568707. 2009-01-23 10:19:27 +01:00
Jan Schmidt
c42c6d6da0 Fix use-after-unref problem noticed by Josep Torra Valles, and run
gst-indent
2009-01-22 22:09:47 +00:00
Wim Taymans
397c00ac33 gst/playback/gstplaybin2.c: Provide the right arguments to a debug line.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (groups_set_locked_state):
Provide the right arguments to a debug line.
2009-01-13 14:47:19 +00:00
Wim Taymans
1e5f963882 gst/playback/gstplaybin2.c: Fix some comments and docs.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
(gst_play_bin_set_uri), (gst_play_bin_set_suburi),
(no_more_pads_cb), (drained_cb), (group_set_locked_state_unlocked),
(activate_group), (deactivate_group), (groups_set_locked_state),
(gst_play_bin_change_state):
Fix some comments and docs.
Post an error message when we fail to link the selector to the sink.
Remove pushing of EOS, this seems unneeded.
Lock the state of deactivated groups so that they don't accidentally
reactivate when the playbin2 state changes.
Reuse uridecodebins.
Unlock and relock state of groups when playbin goes to NULL.
Fixes #566654.
Fixes #566341.
* gst/playback/gsturidecodebin.c: (pad_removed_cb), (type_found):
Only do something in the pad removed callback when we are dealing with
our sourcepads because the sinkpads don't have a ghostpad.
2009-01-07 13:52:14 +00:00
Wim Taymans
8632fc5545 gst/playback/gstplaybin2.c: Disconnect signal handlers before destroying a previous decodebin so that we don't end up...
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (notify_source_cb), (activate_group):
Disconnect signal handlers before destroying a previous decodebin so
that we don't end up causing deadlocks. Fixes #566586.
2009-01-05 12:18:52 +00:00
Wim Taymans
c3ec18af97 gst/audiotestsrc/gstaudiotestsrc.*: Add property to control pull/push based scheduling.
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audio_test_src_class_init), (gst_audio_test_src_init),
(gst_audio_test_src_check_get_range),
(gst_audio_test_src_set_property),
(gst_audio_test_src_get_property):
* gst/audiotestsrc/gstaudiotestsrc.h:
Add property to control pull/push based scheduling.
2009-01-05 10:59:35 +00:00
Wim Taymans
dba6f1f28c gst/playback/gstplaybin2.c: Add some debug info.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (no_more_pads_cb):
Add some debug info.
* gst/playback/gstplaysink.c: (gen_video_chain), (gen_audio_chain),
(gst_play_sink_reconfigure), (gst_play_sink_request_pad),
(gst_play_sink_release_pad):
Add some more debug info.
Reconfigure the audio chain when we switch between raw and encoded audio
in gapless playback.
2008-12-20 12:48:43 +00:00
Stefan Kost
5a30245c38 gst/adder/: Cleanup variable names to make the adder-loop easier to understand.
Original commit message from CVS:
* gst/adder/Makefile.am:
* gst/adder/gstadder.c:
Cleanup variable names to make the adder-loop easier to understand.
Also try to use liboil to spee it up, but ifdef it out as it does not
make any change for me (Intel pentim M (sse,sse2) please try on other
systems).
2008-12-17 08:51:34 +00:00
Wim Taymans
0a6d8f01ef Add minimal docs to make the remaining tcp elements show up.
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/tcp/gsttcpclientsink.c:
* gst/tcp/gsttcpclientsrc.c:
* gst/tcp/gsttcpserversrc.c:
Add minimal docs to make the remaining tcp elements show up.
Fixes #564139.
2008-12-16 20:16:17 +00:00
Stefan Kost
552b5f5fcd gst/playback/: XRef to GstXOverlay.
Original commit message from CVS:
* gst/playback/gstplaybin.c:
* gst/playback/gstplaybin2.c:
XRef to GstXOverlay.
2008-12-12 13:06:48 +00:00
Edward Hervey
89413e390c gst/playback/gsturidecodebin.c: Free the factory array when finalizing.
Original commit message from CVS:
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_finalize):
Free the factory array when finalizing.
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_base_init):
Use a GstStaticPadTemplate since the src pad caps are fixed.
2008-12-12 10:54:45 +00:00
Edward Hervey
7a83664099 gst/subparse/samiparse.c: Some versions of libxml seem to be very picky as to strict formatting of the input and neve...
Original commit message from CVS:
* gst/subparse/samiparse.c: (sami_context_push_state),
(sami_context_pop_state), (start_sami_element), (end_sami_element):
Some versions of libxml seem to be very picky as to strict formatting
of the input and never 'close' the final </body> tag.
In order to fix that bad behaviour, we trigger the flushing of
remaining data on both </body> and </sami>.
Fixes #557365
2008-12-11 15:49:12 +00:00
Guillaume Emont
d477a37e7e gst/typefind/gsttypefindfunctions.c: Add typefinders for MS Word files and OS X .DS_Store files to prevent them to be...
Original commit message from CVS:
Patch by: Guillaume Emont <guillaume at fluendo dot com>
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
Add typefinders for MS Word files and OS X .DS_Store files to
prevent them to be recognized as MPEG files. Fixes bug #564098.
2008-12-11 12:32:03 +00:00
Wim Taymans
08736ec1ae gst/playback/gstplaysink.c: Add some more debug info.
Original commit message from CVS:
* gst/playback/gstplaysink.c: (gen_audio_chain),
(gst_play_sink_reconfigure):
Add some more debug info.
Fix linking of just an encoded sink.
Handle failure to create a sink chain more gracefully than crashing.
2008-12-11 11:04:14 +00:00
Wim Taymans
172e478fd1 gst/playback/gstplaybin2.c: Error out with a missing-plugin error when the input-selector was not found.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (pad_added_cb):
Error out with a missing-plugin error when the input-selector was not
found.
* gst/playback/gstplaysink.c: (gst_play_sink_reconfigure):
Indentation.
2008-12-10 18:43:32 +00:00
Wim Taymans
1bfdc87815 gst/playback/gstplaysink.c: Use G_DEFINE_TYPE.
Original commit message from CVS:
* gst/playback/gstplaysink.c: (gst_play_sink_class_init),
(gst_play_sink_dispose), (gst_play_sink_finalize), (try_element),
(gen_video_chain), (gen_audio_chain), (gst_play_sink_reconfigure),
(gst_play_sink_send_event), (gst_play_sink_change_state):
Use G_DEFINE_TYPE.
Try to set the selected sink to READY before using it. This will allow
for detection of incompatible formats sooner.
Don't cause a fatal error when conversion elements are missing but post
a missing-element message and a warning instead because things might
still link and run fine.
Simplyfy the construction of audio and video sink chains.
2008-12-10 17:39:32 +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
Wim Taymans
cf0efcbff9 gst/volume/gstvolume.c: Use new basetransform vmethod to reconfigure the dynamic properties and any pending volume/mu...
Original commit message from CVS:
* gst/volume/gstvolume.c: (gst_volume_class_init),
(volume_before_transform), (volume_transform_ip):
Use new basetransform vmethod to reconfigure the dynamic properties and
any pending volume/mute changes. Fixes #563508.
2008-12-08 18:44:22 +00:00
Stefan Kost
16e2bccc61 gst/playback/: Add basic docs to decodebin and link to decodebin from decodebin2.
Original commit message from CVS:
* gst/playback/gstdecodebin.c:
* gst/playback/gstdecodebin2.c:
Add basic docs to decodebin and link to decodebin from decodebin2.
2008-12-08 15:25:13 +00:00
Christian Schaller
1048c62b2c fix build
Original commit message from CVS:
fix build
2008-11-28 13:30:36 +00:00
Sebastian Dröge
c514656e3a Update documentation of speexresample for the new element name.
Original commit message from CVS:
* 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/inspect/plugin-videorate.xml:
* gst/speexresample/gstspeexresample.c:
Update documentation of speexresample for the new element name.
2008-11-28 09:44:12 +00:00
Sebastian Dröge
f2eebf3fd3 gst/speexresample/README: Update README with the latest diff between the Speex resampler and our copy.
Original commit message from CVS:
* gst/speexresample/README:
Update README with the latest diff between the Speex resampler
and our copy.
2008-11-28 09:04:46 +00:00
Sebastian Dröge
86144acc9a gst/speexresample/gstspeexresample.c: Update the debug category from speex_resample to audioresample.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c: (plugin_init):
Update the debug category from speex_resample to audioresample.
2008-11-28 08:37:50 +00:00
Sebastian Dröge
7afac6e23a Remove audioresample files.
Original commit message from CVS:
* gst/audioresample/Makefile.am:
* gst/audioresample/buffer.c:
* gst/audioresample/buffer.h:
* gst/audioresample/debug.c:
* gst/audioresample/debug.h:
* gst/audioresample/functable.c:
* gst/audioresample/functable.h:
* gst/audioresample/gstaudioresample.c:
* gst/audioresample/gstaudioresample.h:
* gst/audioresample/resample.c:
* gst/audioresample/resample.h:
* gst/audioresample/resample_chunk.c:
* gst/audioresample/resample_functable.c:
* gst/audioresample/resample_ref.c:
* tests/check/elements/audioresample.c:
Remove audioresample files.
2008-11-27 19:13:59 +00:00
Sebastian Dröge
153406eef5 Rename the moved speexresample to audioresample, integrate into the build system and remove the old audioresample fro...
Original commit message from CVS:
* configure.ac:
* docs/plugins/Makefile.am:
* 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/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-queue2.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* gst/speexresample/gstspeexresample.c: (plugin_init):
* gst/speexresample/Makefile.am:
* tests/check/Makefile.am:
* tests/check/elements/speexresample.c: (setup_speexresample),
(GST_START_TEST), (test_pipeline):
Rename the moved speexresample to audioresample, integrate into the
build system and remove the old audioresample from the build system.
Fixes bug #558124, #385061, #346218, #116051.
2008-11-27 16:57:09 +00:00
Wim Taymans
db785a1f52 gst/playback/gstplaybin2.c: Fix buffer-duration property.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (activate_group):
Fix buffer-duration property.
2008-11-25 11:00:55 +00:00
Michael Smith
aec03a4535 gst/playback/gstplaybin2.c: Add notification of current stream. Add ability to configure buffer sizes.
Original commit message from CVS:
* gst/playback/gstplaybin2.c:
Add notification of current stream. Add ability to configure buffer
sizes.
* gst/playback/gsturidecodebin.c:
Add ability to configure buffer sizes for streaming mode.
Bug #561734.
2008-11-24 20:25:24 +00:00
Jon Trowbridge
0bdeaae59e gst/volume/gstvolume.*: Cleanup volume, define and use default values.
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_choose_func),
(volume_update_volume), (gst_volume_set_volume),
(gst_volume_get_volume), (gst_volume_set_mute),
(gst_volume_class_init), (gst_volume_init),
(volume_process_double), (volume_process_float),
(volume_process_int32), (volume_process_int32_clamp),
(volume_process_int24), (volume_process_int24_clamp),
(volume_process_int16), (volume_process_int16_clamp),
(volume_process_int8), (volume_process_int8_clamp), (volume_setup),
(volume_transform_ip), (volume_set_property),
(volume_get_property):
* gst/volume/gstvolume.h:
Cleanup volume, define and use default values.
Recalculate new volume and mute setup before processing. Fixes #561789.
* tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
Add controller unit test. Patch by: Jonathan Matthew
Fix bogus test that messed with basetransform's internal state.
2008-11-24 12:03:11 +00:00
Wim Taymans
9d0c9fe49b gst/videorate/gstvideorate.c: Add jpeg and png image media types to the caps. Fixes #561436.
Original commit message from CVS:
* gst/videorate/gstvideorate.c:
Add jpeg and png image media types to the caps. Fixes #561436.
2008-11-22 14:44:26 +00:00
Wim Taymans
096efe2fe9 gst/playback/gstplaysink.c: Don't post an error when we can't configure the volume but post a warning instead. Fixes ...
Original commit message from CVS:
* gst/playback/gstplaysink.c: (gen_audio_chain):
Don't post an error when we can't configure the volume but post a
warning instead. Fixes #561780.
2008-11-22 14:31:43 +00:00
Jonathan Rosser
6026260635 gst/videotestsrc/: Add a zone plate pattern generator based on BBC R&D Report 1978/23 (yeah *that* 1978). Try 'video...
Original commit message from CVS:
Patch by: Jonathan Rosser <jonathan.rosser@rd.bbc.co.uk>
* gst/videotestsrc/gstvideotestsrc.c:
* gst/videotestsrc/gstvideotestsrc.h:
* gst/videotestsrc/videotestsrc.c:
* gst/videotestsrc/videotestsrc.h:
Add a zone plate pattern generator based on BBC R&D Report
1978/23 (yeah *that* 1978).  Try 'videotestsrc pattern=zone-plate
kx2=20 ky2=20 kt=1'.
2008-11-21 20:32:56 +00:00
Sebastian Dröge
d36adc543b gst/speexresample/gstspeexresample.c: Add a "filter-length" property that maps to the quality values for compatibilty...
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_class_init), (gst_speex_resample_set_property),
(gst_speex_resample_get_property):
Add a "filter-length" property that maps to the quality values
for compatibilty with audioresample.
2008-11-21 15:45:15 +00:00
Michael Smith
5830b42dc5 gst/playback/gstdecodebin2.c: Fix random fat-fingering making this not compile.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c:
Fix random fat-fingering making this not compile.
2008-11-21 00:04:48 +00:00
Michael Smith
277e46886c gst/playback/gstdecodebin2.c: If the top-level type of the stream is plain text, don't try to decode it, matching beh...
Original commit message from CVS:
* gst/playback/gstdecodebin2.c:
If the top-level type of the stream is plain text, don't try to decode
it, matching behaviour of decodebin.
* gst/playback/gstplaysink.c:
If we fail to generate a text chain (e.g. due to missing optional
plugins), don't crash.
2008-11-20 22:11:38 +00:00
David Schleef
b97e582c57 gst/videotestsrc/: Add "colorspec" property, specifying whether to generate BT.601 or BT.709 video. This only affect...
Original commit message from CVS:
* gst/videotestsrc/gstvideotestsrc.c:
* gst/videotestsrc/gstvideotestsrc.h:
* gst/videotestsrc/videotestsrc.c:
* gst/videotestsrc/videotestsrc.h:
Add "colorspec" property, specifying whether to generate BT.601
or BT.709 video.  This only affects YCbCr values, not RGB, since
if you're generating a 709 test pattern, presumably you want
709 RGB primaries, not 601.  Also add "smpte75" pattern, which
uses 75% colors instead of 100%, since this is often more useful
for testing (and also follows the SMPTE EG-1 guideline).
2008-11-19 00:24:44 +00:00
Alessandro Decina
f39b66e30d gst/playback/gstdecodebin.c: Add a "sink-caps" property to decodebin like it's done for decodebin2.
Original commit message from CVS:
* gst/playback/gstdecodebin.c:
Add a "sink-caps" property to decodebin like it's done for decodebin2.
Fixes #560380.
2008-11-18 18:08:42 +00:00
Jan Schmidt
ca161e799f gst/audioresample/gstaudioresample.c: Guard against a NULL dereference I somehow encountered - with a FLUSH_STOP arri...
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c:
Guard against a NULL dereference I somehow encountered -
with a FLUSH_STOP arriving either before basetransform _start(),
or after _stop().
* gst/typefind/gsttypefindfunctions.c:
Make sure we never jump backwards when typefinding corrupt mov files.
2008-11-14 21:44:33 +00:00
Mark Nauwelaerts
23f10c5403 gst/typefind/gsttypefindfunctions.c: Improve typefinding of ISO JPEG2000 mime types.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (jp2_type_find),
(plugin_init):
Improve typefinding of ISO JPEG2000 mime types.
2008-11-13 21:11:13 +00:00
Wim Taymans
2773fe8f67 gst/playback/gstplaybin2.c: don't try to unlink the selector sinkpad when we don't have it yet. This can happen if an...
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (deactivate_group):
don't try to unlink the selector sinkpad when we don't have it yet. This
can happen if an error occured before the group was complete.
2008-11-13 17:27:37 +00:00
Wim Taymans
79199d884f gst/playback/gstplaybin2.c: Catch state change errors and stop from the uridecodebin elements instead of trying to co...
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (activate_group):
Catch state change errors and stop from the uridecodebin elements
instead of trying to continue in vain.
2008-11-11 15:52:14 +00:00
Thomas Vander Stichele
9b6f3ad0c8 gst/adder/gstadder.c: Change author string after seeing output of gst-inspector.
Original commit message from CVS:
* gst/adder/gstadder.c:
Change author string after seeing output of gst-inspector.
2008-11-10 13:55:08 +00:00
Wim Taymans
9045e0428e gst/playback/gstplaysink.c: Don't try to do crazy things when we only have a text pad without a video pad. Fixes #559...
Original commit message from CVS:
* gst/playback/gstplaysink.c: (gst_play_sink_reconfigure):
Don't try to do crazy things when we only have a text pad without a
video pad. Fixes #559478.
2008-11-10 10:33:26 +00:00
Wim Taymans
8e07d4ec69 gst/volume/gstvolume.*: Keep negotiated state in a separate variable.
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_update_real_volume),
(gst_volume_set_volume), (gst_volume_get_volume),
(gst_volume_set_mute), (gst_volume_init), (volume_setup),
(volume_transform_ip), (volume_update_mute),
(volume_update_volume), (volume_get_property):
* gst/volume/gstvolume.h:
Keep negotiated state in a separate variable.
Protect the volume and mute properties with the object lock.
Protect modifying the transform with the transform lock.
2008-11-05 19:18:25 +00:00
Wim Taymans
d9ae8db094 gst/ffmpegcolorspace/gstffmpegcodecmap.c: Only convert caps to string when debug is enabled.
Original commit message from CVS:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_pixfmt_to_caps):
Only convert caps to string when debug is enabled.
2008-11-05 12:20:21 +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
Sebastian Dröge
35acee194f gst/speexresample/gstspeexresample.c: Add TODO at the top of the file for enabling SSE/ARM specific optimizations and...
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_convert_buffer):
Add TODO at the top of the file for enabling SSE/ARM specific
optimizations and choosing the fastest implementation at runtime.
Add g_assert_not_reached() at two places that should really never
be reached.
2008-11-03 08:55:49 +00:00
Sebastian Dröge
9c88e06594 gst/speexresample/gstspeexresample.c: Fix format string and arguments.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_check_discont):
Fix format string and arguments.
* gst/speexresample/resample_sse.h:
Add missing file.
2008-11-02 09:19:24 +00:00
Sebastian Dröge
fa613940ef gst/speexresample/: Add missing headers to Makefile.am.
Original commit message from CVS:
* gst/speexresample/Makefile.am:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_base_init), (gst_speex_resample_get_funcs),
(gst_speex_resample_convert_buffer), (_benchmark_int_float),
(_benchmark_int_int), (_benchmark_integer_resampling),
(plugin_init):
* gst/speexresample/gstspeexresample.h:
* gst/speexresample/resample.c:
* gst/speexresample/speex_resampler_double.c:
* gst/speexresample/speex_resampler_float.c:
* gst/speexresample/speex_resampler_int.c:
* gst/speexresample/speex_resampler_wrapper.h:
Add missing headers to Makefile.am.
Update copyright, years and my mail address.
Benchmark the integer resampling implementation against the
float implementation and use the faster one for 8/16 bit integer
input. On most recent systems the floating point version is faster.
2008-11-01 19:38:36 +00:00
Sebastian Dröge
35a5ad819d gst/speexresample/gstspeexresample.c: The length for the buffer conversion function is the number of audio frames, i....
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_convert_buffer):
The length for the buffer conversion function is the number of
audio frames, i.e. we need to multiply it by the number of channels
to get the number of values. Also spotted by the unit test after
running in valgrind.
2008-10-30 14:55:43 +00:00
Sebastian Dröge
b2cbf8f91d tests/check/elements/speexresample.c: Add pipeline unit tests for testing all supported formats with up/downsampling ...
Original commit message from CVS:
* tests/check/elements/speexresample.c: (element_message_cb),
(eos_message_cb), (test_pipeline), (GST_START_TEST),
(speexresample_suite):
Add pipeline unit tests for testing all supported formats with
up/downsampling and different in/outrates.
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_push_drain), (gst_speex_resample_process):
* gst/speexresample/speex_resampler_wrapper.h:
Fix bugs identified by the testsuite.
2008-10-30 14:46:31 +00:00
Sebastian Dröge
4681a87cce gst/speexresample/: Add support for int8, int24 and int32 input by converting internally to/from int16 or double.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c: (gst_speex_resample_stop),
(gst_speex_resample_get_funcs),
(gst_speex_resample_transform_size),
(gst_speex_resample_convert_buffer),
(gst_speex_resample_push_drain), (gst_speex_resample_process):
* gst/speexresample/gstspeexresample.h:
* gst/speexresample/speex_resampler_wrapper.h:
Add support for int8, int24 and int32 input by converting internally
to/from int16 or double.
2008-10-30 13:44:41 +00:00
Sebastian Dröge
d80b5c4aae Add support for double samples as input and refactor the usage of the different compilation flavors of the speex resa...
Original commit message from CVS:
* gst/speexresample/Makefile.am:
* gst/speexresample/arch.h:
* gst/speexresample/gstspeexresample.c: (gst_speex_resample_stop),
(gst_speex_resample_get_unit_size), (gst_speex_resample_get_funcs),
(gst_speex_resample_init_state), (gst_speex_resample_update_state),
(gst_speex_resample_reset_state), (gst_speex_resample_parse_caps),
(_gcd), (gst_speex_resample_transform_size),
(gst_speex_resample_set_caps), (gst_speex_resample_push_drain),
(gst_speex_resample_process), (gst_speex_resample_transform),
(gst_speex_resample_query), (gst_speex_resample_set_property):
* gst/speexresample/gstspeexresample.h:
* gst/speexresample/resample.c:
* gst/speexresample/speex_resampler.h:
* gst/speexresample/speex_resampler_double.c:
* gst/speexresample/speex_resampler_wrapper.h:
* tests/check/elements/speexresample.c: (setup_speexresample),
(test_perfect_stream_instance), (GST_START_TEST),
(test_discont_stream_instance):
Add support for double samples as input and refactor the usage
of the different compilation flavors of the speex resampler.
2008-10-30 12:43:44 +00:00
Stefan Kost
087676f09b gst/audioresample/gstaudioresample.c: Return the result of parent_class->event().
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c:
Return the result of parent_class->event().
2008-10-30 11:43:12 +00:00
Sebastian Dröge
f5b4fa17ff gst/speexresample/gstspeexresample.*: Rewrite timestamp tracking to make it more robust and guarantee a continous str...
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c: (gst_speex_resample_start),
(gst_speex_resample_get_unit_size),
(gst_speex_resample_push_drain), (gst_speex_resample_event),
(gst_speex_resample_check_discont), (gst_speex_resample_process),
(gst_speex_resample_transform):
* gst/speexresample/gstspeexresample.h:
Rewrite timestamp tracking to make it more robust and guarantee
a continous stream.
* tests/check/Makefile.am:
* tests/check/elements/speexresample.c: (setup_speexresample),
(cleanup_speexresample), (fail_unless_perfect_stream),
(test_perfect_stream_instance), (GST_START_TEST),
(test_discont_stream_instance), (live_switch_alloc_only_48000),
(live_switch_get_sink_caps), (live_switch_push),
(speexresample_suite):
Add unit tests for speexresample based on the audioresample unit tests.
2008-10-29 12:11:20 +00:00
Sebastian Dröge
305b2f3d14 gst/speexresample/gstspeexresample.*: Some random cleanup, add G_LIKELY and friends, use GST_DEBUG_OBJECT instead of ...
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_get_unit_size),
(gst_speex_resample_fixate_caps), (gst_speex_resample_init_state),
(gst_speex_resample_update_state), (gst_speex_resample_parse_caps),
(gst_speex_resample_transform_size), (gst_speex_resample_set_caps),
(gst_speex_resample_push_drain), (gst_speex_resample_event),
(gst_speex_resample_check_discont), (gst_speex_fix_output_buffer),
(gst_speex_resample_process), (gst_speex_resample_transform),
(gst_speex_resample_query), (gst_speex_resample_set_property):
* gst/speexresample/gstspeexresample.h:
Some random cleanup, add G_LIKELY and friends, use GST_DEBUG_OBJECT
instead of GST_DEBUG, ...
2008-10-28 19:30:33 +00:00
Sebastian Dröge
6009490cef gst/speexresample/gstspeexresample.c: Fixate to the nearest supported rate instead of the first one.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_class_init), (gst_speex_resample_fixate_caps),
(gst_speex_resample_process):
Fixate to the nearest supported rate instead of the first one.
2008-10-28 16:28:45 +00:00
Sebastian Dröge
70348d7327 gst/audioresample/gstaudioresample.c: Fixate the rate to the nearest supported rate instead of the first one. Fixes b...
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c:
(gst_audioresample_class_init), (audioresample_fixate_caps):
Fixate the rate to the nearest supported rate instead of
the first one. Fixes bug #549510.
2008-10-28 16:25:00 +00:00
Sebastian Dröge
08489d15ed gst/speexresample/: Update Speex resampler with latest version from Speex GIT.
Original commit message from CVS:
* gst/speexresample/README:
* gst/speexresample/arch.h:
* gst/speexresample/fixed_arm4.h:
* gst/speexresample/fixed_arm5e.h:
* gst/speexresample/fixed_bfin.h:
* gst/speexresample/fixed_debug.h:
* gst/speexresample/fixed_generic.h:
* gst/speexresample/resample.c: (compute_func), (main), (sinc),
(cubic_coef), (resampler_basic_direct_single),
(resampler_basic_direct_double),
(resampler_basic_interpolate_single),
(resampler_basic_interpolate_double), (update_filter),
(speex_resampler_init_frac), (speex_resampler_process_native),
(speex_resampler_magic), (speex_resampler_process_float),
(speex_resampler_process_int),
(speex_resampler_process_interleaved_float),
(speex_resampler_process_interleaved_int),
(speex_resampler_set_rate_frac), (speex_resampler_skip_zeros),
(speex_resampler_reset_mem):
* gst/speexresample/speex_resampler.h:
Update Speex resampler with latest version from Speex GIT.
2008-10-28 11:46:28 +00:00
Sebastian Dröge
e4e86b0bad gst/typefind/gsttypefindfunctions.c: Improve MXF typefinding a bit by searching for a header partition pack instead o...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mxf_type_find):
Improve MXF typefinding a bit by searching for a header partition
pack instead of just a general partition pack and checking more
bytes for valid values.
2008-10-20 14:08:52 +00:00
Stefan Kost
2cd4c7e2b9 Don't install static libs for plugins. Fixes #550851 for base.
Original commit message from CVS:
* ext/alsa/Makefile.am:
* ext/cdparanoia/Makefile.am:
* ext/gio/Makefile.am:
* ext/gnomevfs/Makefile.am:
* ext/libvisual/Makefile.am:
* ext/ogg/Makefile.am:
* ext/pango/Makefile.am:
* ext/theora/Makefile.am:
* ext/vorbis/Makefile.am:
* gst/adder/Makefile.am:
* gst/audioconvert/Makefile.am:
* gst/audiorate/Makefile.am:
* gst/audioresample/Makefile.am:
* gst/audiotestsrc/Makefile.am:
* gst/ffmpegcolorspace/Makefile.am:
* gst/gdp/Makefile.am:
* gst/playback/Makefile.am:
* gst/subparse/Makefile.am:
* gst/tcp/Makefile.am:
* gst/typefind/Makefile.am:
* gst/videorate/Makefile.am:
* gst/videoscale/Makefile.am:
* gst/videotestsrc/Makefile.am:
* gst/volume/Makefile.am:
* sys/v4l/Makefile.am:
* sys/ximage/Makefile.am:
* sys/xvimage/Makefile.am:
Don't install static libs for plugins. Fixes #550851 for base.
2008-10-16 15:07:00 +00:00
Wim Taymans
5ad1ebcf4c gst/audiotestsrc/gstaudiotestsrc.c: Set the default blocksize to -1 because we will then use the configured samplespe...
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_init):
Set the default blocksize to -1 because we will then use the configured
samplesperbuffer to create our output buffer.
2008-10-16 13:50:00 +00:00
Edward Hervey
e68dbb884d gst/typefind/gsttypefindfunctions.c: Don't forget to advance the offset of what we're matching against, else we end u...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (flac_type_find):
Don't forget to advance the offset of what we're matching against, else
we end up in a forever loop.
2008-10-15 14:25:50 +00:00
Sebastian Dröge
e86d1dd432 gst/subparse/gstsubparse.c: Improve typefinding a bit. If we don't have a Unicode charset try GST_SUBTITLE_ENCODING a...
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (gst_subparse_type_find):
Improve typefinding a bit. If we don't have a Unicode charset
try GST_SUBTITLE_ENCODING and otherwise try ISO-8859-15.
2008-10-15 11:25:09 +00:00
Sebastian Dröge
e7b42af896 gst/subparse/: Add support for UTF16/UTF32 subtitles as long as the first bytes of the first buffer contain the BOM. ...
Original commit message from CVS:
* gst/subparse/Makefile.am:
* gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
(gst_sub_parse_class_init), (gst_sub_parse_init),
(gst_convert_to_utf8), (detect_encoding), (convert_encoding),
(get_next_line), (gst_sub_parse_data_format_autodetect),
(feed_textbuf), (handle_buffer), (gst_sub_parse_change_state),
(gst_subparse_type_find):
* gst/subparse/gstsubparse.h:
Add support for UTF16/UTF32 subtitles as long as the first bytes of
the first buffer contain the BOM. This also adds support for other
encodings that allow NUL bytes via the encoding property.
Fixes bugs #552237 and #456788.
2008-10-13 08:58:29 +00:00
Sebastian Dröge
d0d588ff6f gst/typefind/gsttypefindfunctions.c: For looking at the 4th byte we have to get 4 bytes of course and not 3.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (flac_type_find):
For looking at the 4th byte we have to get 4 bytes of course
and not 3.
2008-10-13 08:00:55 +00:00
Sebastian Dröge
862fd1d50f gst/typefind/gsttypefindfunctions.c: Improve FLAC-without-headers typefinding by looking at most of the frame header ...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (flac_type_find):
Improve FLAC-without-headers typefinding by looking at most of the
frame header and checking if invalid values are used. Should prevent
quite some false positives compared to the old version which only
check if the first 14 bits are set.
2008-10-13 07:52:41 +00:00
Sebastian Dröge
60bf63486b Add support for subtitle files with UTF-8 BOM at the beginning by simple stripping it from the first line before pass...
Original commit message from CVS:
* gst/subparse/gstsubparse.c:
(gst_sub_parse_data_format_autodetect), (handle_buffer),
(gst_sub_parse_change_state):
* gst/subparse/gstsubparse.h:
* tests/check/elements/subparse.c: (GST_START_TEST):
Add support for subtitle files with UTF-8 BOM at the beginning
by simple stripping it from the first line before passing it
to any parsing code. Fixes bug #555257 and playback of files
created by Gnome Subtitles.
2008-10-10 17:13:40 +00:00
Wim Taymans
81f5117fa9 gst/audiotestsrc/gstaudiotestsrc.*: Define the default property values in the usual place.
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audio_test_src_class_init), (gst_audio_test_src_init),
(gst_audio_test_src_src_fixate), (gst_audio_test_src_setcaps),
(gst_audio_test_src_start), (gst_audio_test_src_stop),
(gst_audio_test_src_do_seek), (gst_audio_test_src_check_get_range),
(gst_audio_test_src_create):
* gst/audiotestsrc/gstaudiotestsrc.h:
Define the default property values in the usual place.
Implement start/stop to reset values correctly.
Calculate the sample size only once when we negotiate.
Rename some values to make more sense.
Keep track of our byte range.
Add support for pull based scheduling. Disabled for now until we have
the whole stack working.
Set the BUFFER_OFFSET correctly.
2008-10-10 15:45:15 +00:00
Sebastian Dröge
b735321f58 Make the detection of the used subtitle a bit less strict for srt subtitles. Fixes bug #555607.
Original commit message from CVS:
Based on a patch by: xavierb at gmail dot com
* gst/subparse/gstsubparse.c:
(gst_sub_parse_data_format_autodetect):
* tests/check/elements/subparse.c: (GST_START_TEST):
Make the detection of the used subtitle a bit less strict
for srt subtitles. Fixes bug #555607.
2008-10-10 15:32:10 +00:00
Wim Taymans
fbeec41546 gst/playback/gstdecodebin2.c: Remove bogus assert, the decodepad could have been created inside an already existing g...
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_pad_activate):
Remove bogus assert, the decodepad could have been created inside an
already existing group.
2008-10-08 14:44:04 +00:00
Andy Wingo
3329abde33 gst/playback/gstdecodebin2.c (expose_pad): Fix typo: unset target instead of setting it.
Original commit message from CVS:
2008-10-08  Andy Wingo  <wingo@pobox.com>

* gst/playback/gstdecodebin2.c (expose_pad): Fix typo: unset
target instead of setting it.
(gst_decode_pad_activate, gst_decode_pad_unblock): This is now the
API for a decode pad. The bugfix is that we set the group in
activate(), not when the pad was created because it might be NULL
then.
(gst_decode_group_control_source_pad, gst_decode_group_expose):
Update to use the API.
2008-10-08 14:00:07 +00:00
Andy Wingo
6c7e1c8a9b gst/playback/gstdecodebin2.c (struct _GstDecodePad): Change to be a subclass of GstGhostPad.
Original commit message from CVS:
2008-10-08  Andy Wingo  <wingo@pobox.com>

* gst/playback/gstdecodebin2.c (struct _GstDecodePad): Change to
be a subclass of GstGhostPad.
(analyze_new_pad): So, when emitting the signals that determine
how we do autoplugging, already create the ghost pad and use it as
the pad in the signal arguments. This allows applications to make
a connection between the pad passed in e.g. autoplug-continue, and
the pad passed in new-decoded-pad.
(connect_pad, expose_pad): Update to receive the ghosted decode
pad in the args, retargetting it as necessary if we have to plug
the target pad through a multiqueue.
(gst_decode_group_control_source_pad): Adapt to receive an
already-ghosted pad that just needs activation, blocking, and
drain notification.
(sort_end_pads): Adapt for decode pads actually being pads.
(gst_decode_group_expose): Adapt for decode pads actually being
pads. Rewrite the decode pad names so they appear in order. Adds a
new error case if we couldn't set the name.
(gst_decode_group_free, gst_decode_group_hide): Adapt cleanup
logic.
(gst_decode_pad_set_blocked, gst_decode_pad_add_drained_check):
New API for the decode pad, needed because we shouldn't do these
things inside gst_decode_pad_new(), but after.
(gst_decode_pad_new): Change to actually make the real pad, and
delay the blocking/drainage bits.
2008-10-08 12:49:40 +00:00
Sebastian Dröge
c915582c17 gst/audioconvert/audioconvert.c: Prevent overflows with big buffer when calculating the size of the intermediate buff...
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (audio_convert_convert):
Prevent overflows with big buffer when calculating the size of
the intermediate buffer by using gst_util_uint64_scale() instead of
plain arithmetics. Fixes bug #552801.
2008-10-08 11:50:50 +00:00
Sebastian Dröge
44143f1dcc gst/typefind/gsttypefindfunctions.c: Add typefinder for MXF.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mxf_type_find),
(plugin_init):
Add typefinder for MXF.
2008-10-05 08:11:53 +00:00
Sebastian Dröge
d3edbe7745 gst/typefind/gsttypefindfunctions.c: Add typefinder for MXF.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mxf_type_find),
(plugin_init):
Add typefinder for MXF.
2008-10-05 08:10:09 +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
Mark Nauwelaerts
ec6afbd321 gst/typefind/gsttypefindfunctions.c: Typefind video/mj2 and image/jp2 ISO JPEG2000 mime types.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
Typefind video/mj2 and image/jp2 ISO JPEG2000 mime types.
Fixes #550638.
2008-09-03 12:23:44 +00:00
Stefan Kost
1875564b65 Rework last change, so that we build subparse, but just disable the sami parse functionality, if we're configured to ...
Original commit message from CVS:
* configure.ac:
* gst/subparse/Makefile.am:
* gst/subparse/gstsubparse.c:
* gst/subparse/samiparse.c:
* tests/check/elements/subparse.c:
Rework last change, so that we build subparse, but just disable the
sami parse functionality, if we're configured to not use xml. In the
tests only the sami test is disabled now.
2008-09-03 10:12:04 +00:00
Jonathan Matthew
686a893a0f gst/typefind/gsttypefindfunctions.c: Add typefinder for PDF documents (which is nice to have, since it's a common for...
Original commit message from CVS:
Patch by: Jonathan Matthew  <notverysmart gmail com>
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
Add typefinder for PDF documents (which is nice to have, since it's a
common format, but also helps prevent false positives). Fixes #549814.
2008-08-30 15:55:06 +00:00
Wim Taymans
ed11048c05 gst/playback/gstplaybin2.c: Fix nasty race where multiple decodebins could start pushing data before we manage to con...
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (selector_blocked), (pad_added_cb),
(no_more_pads_cb):
Fix nasty race where multiple decodebins could start pushing data before
we manage to configure the sinks, resulting in not-linked errors in
typical RTSP streaming cases.
2008-08-27 15:30:16 +00:00
David Schleef
7cce52603e gst/typefind/gsttypefindfunctions.c: DV typefinding. Remove check for a bit that is 0 in IEC 61384, but not SMPTE 314M.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: DV typefinding.  Remove
check for a bit that is 0 in IEC 61384, but not SMPTE 314M.
Fixes #548065.
2008-08-16 20:57:27 +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
Andy Wingo
79930b61bf gst/audiotestsrc/gstaudiotestsrc.c: Very crucial and important documentation fix.
Original commit message from CVS:
2008-08-04  Andy Wingo  <wingo@pobox.com>

* gst/audiotestsrc/gstaudiotestsrc.c: Very crucial and important
documentation fix.
2008-08-04 09:11:08 +00:00
Stefan Kost
7c2a26c9ed gst/adder/gstadder.c: Cleanup lots of empty lines that came from gst-indent going havoc before I added the INDENT_ON/...
Original commit message from CVS:
* gst/adder/gstadder.c:
Cleanup lots of empty lines that came from gst-indent going havoc
before I added the INDENT_ON/OFF marker some time agao.
2008-08-01 13:06:59 +00:00
Wim Taymans
76456cb647 gst/playback/gstplaysink.c: Add some more comments.
Original commit message from CVS:
* gst/playback/gstplaysink.c: (gst_play_sink_set_vis_plugin):
Add some more comments.
2008-07-31 13:06:13 +00:00
Wim Taymans
824a8fc80c gst/videotestsrc/gstvideotestsrc.c: Discard buffers of the wrong size after renegotiation, this is perfectly possible...
Original commit message from CVS:
* gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_getcaps),
(gst_video_test_src_create):
Discard buffers of the wrong size after renegotiation, this is perfectly
possible with things like capsfilter that could suggest caps changes
upstream without knowing the size of the buffer.
2008-07-31 12:58:44 +00:00
Tim-Philipp Müller
58c48279dc gst/audioconvert/Makefile.am: Dist recently-added gstfastrandom.h.
Original commit message from CVS:
* gst/audioconvert/Makefile.am:
Dist recently-added gstfastrandom.h.
2008-07-30 19:51:36 +00:00
Stefan Kost
feea3e0b1c gst/playback/gstplaybasebin.c: Fix property doc markup (its not a signal).
Original commit message from CVS:
* gst/playback/gstplaybasebin.c:
Fix property doc markup (its not a signal).
* sys/xvimage/xvimagesink.c:
Add since tag for new proeprties (also add sice tags fro the last two
other additions).
2008-07-29 10:26:28 +00:00
Sebastian Dröge
63b89f5625 gst/typefind/gsttypefindfunctions.c: Add simple typefinder for the CELT codec (www.celt-codec.org).
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (celt_type_find),
(plugin_init):
Add simple typefinder for the CELT codec (www.celt-codec.org).
2008-07-28 12:47:06 +00:00
Sebastian Dröge
ef5004e56e gst/audioconvert/: Implement a linear congruential generator as pseudo random number generator for the dither noise. ...
Original commit message from CVS:
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioquantize.c:
(gst_audio_quantize_setup_dither),
(gst_audio_quantize_free_dither):
* gst/audioconvert/gstfastrandom.h:
Implement a linear congruential generator as pseudo random number
generator for the dither noise. This is about 2 times faster than
using GLib's mersenne twister. Also this uses only integer math for
generating integers while GLib internally uses floating point math.
2008-07-23 18:34:19 +00:00
David Schleef
cc74285d12 gst/audiotestsrc/gstaudiotestsrc.*: Add 'ticks', a 1/30 second sine wave pulse every second.
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
* gst/audiotestsrc/gstaudiotestsrc.h:
Add 'ticks', a 1/30 second sine wave pulse every second.
2008-07-17 02:30:24 +00:00
Jan Schmidt
024d0e56f5 gst/playback/: First stab at integrating DVD subpicture overlay into playbin. Successfully plugs and plays, but the q...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c:
* gst/playback/gstplaybasebin.h:
* gst/playback/gstplaybin.c:
* gst/playback/gststreamselector.c:
First stab at integrating DVD subpicture overlay into
playbin. Successfully plugs and plays, but the queues need
shrinking - 3 seconds of video is too much buffering.
2008-07-14 08:18:58 +00:00
Stefan Kost
8b24a3a057 gst/audioconvert/gstaudioconvert.c: Remove now obsolete note in the docs.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
Remove now obsolete note in the docs.
2008-07-11 18:06:33 +00:00
Stefan Kost
4f699b7f80 Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipe...
Original commit message from CVS:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-overrides.txt:
* 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-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-queue2.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* ext/alsa/gstalsamixer.c:
* ext/alsa/gstalsasink.c:
* ext/alsa/gstalsasrc.c:
* ext/gio/gstgiosink.c:
* ext/gio/gstgiosrc.c:
* ext/gio/gstgiostreamsink.c:
* ext/gio/gstgiostreamsrc.c:
* ext/gnomevfs/gstgnomevfssink.c:
* ext/gnomevfs/gstgnomevfssrc.c:
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggmux.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/theora/theoraparse.c:
* ext/vorbis/vorbisdec.c:
* ext/vorbis/vorbisenc.c:
* ext/vorbis/vorbisparse.c:
* ext/vorbis/vorbistag.c:
* gst/adder/gstadder.c:
* gst/audioconvert/gstaudioconvert.c:
* gst/audioresample/gstaudioresample.c:
* gst/audiotestsrc/gstaudiotestsrc.c:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/gdp/gstgdpdepay.c:
* gst/gdp/gstgdppay.c:
* gst/playback/gstdecodebin2.c:
* gst/playback/gstplaybin.c:
* gst/playback/gstplaybin2.c:
* gst/playback/gstqueue2.c:
* gst/playback/gsturidecodebin.c:
* gst/tcp/gstmultifdsink.c:
* gst/tcp/gsttcpserversink.c:
* gst/videorate/gstvideorate.c:
* gst/videoscale/gstvideoscale.c:
* gst/videotestsrc/gstvideotestsrc.c:
* gst/volume/gstvolume.c:
* sys/ximage/ximagesink.c:
* sys/xvimage/xvimagesink.c:
Cleanup Plugin docs. Link to signals and properties. Fix sub-section
titles. Drop mentining that all our example pipelines are "simple"
pipelines.
2008-07-11 06:10:24 +00:00
Stefan Kost
2b33c755b6 Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipe...
Original commit message from CVS:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-overrides.txt:
* 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-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-queue2.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* ext/alsa/gstalsamixer.c:
* ext/alsa/gstalsasink.c:
* ext/alsa/gstalsasrc.c:
* ext/gio/gstgiosink.c:
* ext/gio/gstgiosrc.c:
* ext/gio/gstgiostreamsink.c:
* ext/gio/gstgiostreamsrc.c:
* ext/gnomevfs/gstgnomevfssink.c:
* ext/gnomevfs/gstgnomevfssrc.c:
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggmux.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/theora/theoraparse.c:
* ext/vorbis/vorbisdec.c:
* ext/vorbis/vorbisenc.c:
* ext/vorbis/vorbisparse.c:
* ext/vorbis/vorbistag.c:
* gst/adder/gstadder.c:
* gst/audioconvert/gstaudioconvert.c:
* gst/audioresample/gstaudioresample.c:
* gst/audiotestsrc/gstaudiotestsrc.c:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/gdp/gstgdpdepay.c:
* gst/gdp/gstgdppay.c:
* gst/playback/gstdecodebin2.c:
* gst/playback/gstplaybin.c:
* gst/playback/gstplaybin2.c:
* gst/playback/gstqueue2.c:
* gst/playback/gsturidecodebin.c:
* gst/tcp/gstmultifdsink.c:
* gst/tcp/gsttcpserversink.c:
* gst/videorate/gstvideorate.c:
* gst/videoscale/gstvideoscale.c:
* gst/videotestsrc/gstvideotestsrc.c:
* gst/volume/gstvolume.c:
* sys/ximage/ximagesink.c:
* sys/xvimage/xvimagesink.c:
Cleanup Plugin docs. Link to signals and properties. Fix sub-section
titles. Drop mentining that all our example pipelines are "simple"
pipelines.
2008-07-10 21:06:06 +00:00
Sebastian Dröge
b02dc1bf6a gst/playback/gstdecodebin.c: And ref the pad before returning it again when linking to the queue failed. Otherwise we...
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (add_raw_queue):
And ref the pad before returning it again when linking to the queue
failed. Otherwise we will unref the pad twice later and things break.
2008-07-07 09:55:41 +00:00
Sebastian Dröge
ba9c438f98 gst/playback/gstdecodebin.c: If linking the raw pad with a queue fails, try it without a queue instead of failing com...
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (add_raw_queue):
If linking the raw pad with a queue fails, try it without a queue
instead of failing completely. This should never happen.
2008-07-07 09:48:45 +00:00
Evgeniy Stepanov
bddd224b36 gst/playback/gstdecodebin.c: Add a queue after a demuxer if the demuxer outputs raw data. This was done before only f...
Original commit message from CVS:
Patch by: Evgeniy Stepanov <eugeni dot stepanov at gmail dot com>
* gst/playback/gstdecodebin.c: (add_raw_queue), (close_pad_link):
Add a queue after a demuxer if the demuxer outputs raw data. This was
done before only for non-raw data but is required in this case too.
Fixes bug #540215.
decodebin2 doesn't have this issue because all streams of a group
go through multiqueue.
2008-07-06 23:22:12 +00:00
Wim Taymans
3fb8f3b0dd gst/videotestsrc/gstvideotestsrc.*: Cleanups, use default property values as defines.
Original commit message from CVS:
* gst/videotestsrc/gstvideotestsrc.c:
(gst_video_test_src_class_init), (gst_video_test_src_init),
(gst_video_test_src_set_property),
(gst_video_test_src_get_property), (gst_video_test_src_create):
* gst/videotestsrc/gstvideotestsrc.h:
Cleanups, use default property values as defines.
Add property to enable/disable peer buffer allocation.
2008-07-01 13:22:49 +00:00
Sebastian Dröge
a97dc76ad7 gst/ffmpegcolorspace/: Only set/get on the PAL8 format, ffmpegcolorspace doesn't support it on other formats. Also ad...
Original commit message from CVS:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_pixfmt_to_caps):
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_get_unit_size):
Only set/get on the PAL8 format, ffmpegcolorspace doesn't support
it on other formats. Also adjust the unit size only for that format
to not include the palette. Fixes bug #540497.
2008-06-30 08:29:09 +00:00
Stefan Kost
e0d27d23cc gst/adder/gstadder.c: Use GST_DEBUG_FUNCPTR and remove some extra vlnak lines.
Original commit message from CVS:
* gst/adder/gstadder.c:
Use GST_DEBUG_FUNCPTR and remove some extra vlnak lines.
2008-06-29 13:45:27 +00:00
Stefan Kost
69f2aaea3c gst/playback/gstqueue2.c: Do not double notify. Remove the unsued return value.
Original commit message from CVS:
* gst/playback/gstqueue2.c:
Do not double notify. Remove the unsued return value.
2008-06-24 16:22:45 +00:00
Michael Smith
6ef8ecd7a3 gst/playback/: Add get-video-pad, get-audio-pad, get-text-pad action signals to playbin2. This allows the user to get...
Original commit message from CVS:
* gst/playback/gstplay-marshal.list:
* gst/playback/gstplaybin2.c:
Add get-video-pad, get-audio-pad, get-text-pad action signals to
playbin2. This allows the user to get to the selector's sinkpads, and
thus inspect a range of things - caps, tags, etc.
2008-06-20 18:24:24 +00:00
Michael Smith
f5b9e8c065 gst/playback/gstplaybin2.c: Use a different constant for the convert-frame signal id.
Original commit message from CVS:
* gst/playback/gstplaybin2.c:
Use a different constant for the convert-frame signal id.
Fixes #537009.
2008-06-20 17:27:03 +00:00
Michael Smith
8a59d948d9 gst/playback/: Fix a whole bunch of typos in comments and log statements.
Original commit message from CVS:
* gst/playback/gstplaybin2.c:
* gst/playback/gstplaysink.c:
Fix a whole bunch of typos in comments and log statements.
2008-06-20 17:18:55 +00:00
Michael Smith
9d2564874a gst/playback/gstdecodebin2.c: Ensure decodebin2 emits 'drained' signal once, and only once, when all pads are drained.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c:
Ensure decodebin2 emits 'drained' signal once, and only once, when all
pads are drained.
2008-06-20 16:56:18 +00:00
Thomas Vander Stichele
bcc3b3b5f5 apparently it's an error to specify nc -l -p 3000 - though the short usage does not make it very clear that you can d...
Original commit message from CVS:
apparently it's an error to specify nc -l -p 3000 - though the short usage
does not make it very clear that you can drop the host arg with -l
2008-06-20 16:12:50 +00:00
Wim Taymans
cf7da52701 gst/playback/gstplaybin2.c: Implement the source property, emit notify when it changes in the underlying uridecodebin.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (gst_play_bin_get_property),
(notify_source), (activate_group):
Implement the source property, emit notify when it changes in the
underlying uridecodebin.
2008-06-20 09:19:59 +00:00
Antoine Tremblay
1a71c15677 gst/gdp/gstgdppay.c: Fix a buffer memleak and remove a confusing and wrong debug output.
Original commit message from CVS:
Patch by: Antoine Tremblay <hexa00 at gmail dot com>
* gst/gdp/gstgdppay.c: (gst_gdp_pay_reset):
Fix a buffer memleak and remove a confusing and wrong debug output.
Fixes bug #538663.
2008-06-20 08:45:13 +00:00
Stefan Kost
332fe99892 Final round of doc updates.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/speed/gstspeed.c:
* gst/speexresample/gstspeexresample.c:
* gst/videosignal/gstvideoanalyse.c:
* gst/videosignal/gstvideodetect.c:
* gst/videosignal/gstvideomark.c:
* sys/dvb/gstdvbsrc.c:
* sys/oss4/oss4-mixer.c:
* sys/oss4/oss4-sink.c:
* sys/oss4/oss4-source.c:
* sys/wininet/gstwininetsrc.c:
Final round of doc updates.
2008-06-16 07:30:32 +00:00
Michael Smith
2fdd607e95 gst/playback/gstplaybasebin.c: Disconnect signals from decodebins we created before we remove it from playbin, to avo...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c:
Disconnect signals from decodebins we created before we remove it from
playbin, to avoid crashes if the decodebin is eventually disposed after
the playbin itself (possible if the app takes a reference on the
decodebin).
Fixes #536521.
2008-06-04 17:42:38 +00:00
Tim-Philipp Müller
93db55c074 gst/typefind/gsttypefindfunctions.c: Bunch of small clean-ups: use gst_type_find_suggest_simple(); don't copy caps fo...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (aac_type_find),
(mp3_type_find), (musepack_type_find), (MULTIPART_MAX_HEADER_SIZE),
(mpeg_sys_type_find), (mpeg_ts_type_find), (mpeg4_video_type_find),
(h264_video_type_find), (mpeg_video_stream_type_find),
(dv_type_find), (mmsh_type_find):
Bunch of small clean-ups: use gst_type_find_suggest_simple(); don't
copy caps for no good reason (this may be desirable to make it easier
to detect leaks, but then it should probably be done for all caps
in the typefinder somewhere).
2008-06-04 17:12:40 +00:00
Peter Kjellerstedt
c140528357 gst/tcp/gstmultifdsink.c: Fixed accidental use of IPv4 options for all IPv6 addresses.
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (setup_dscp_client):
Fixed accidental use of IPv4 options for all IPv6 addresses.
2008-06-04 11:33:23 +00:00
Antoine Tremblay
be2f6a8085 gst/gdp/gstgdppay.c: Don't set caps on the buffers that contain a copy of the buffer including the caps of them resul...
Original commit message from CVS:
Patch by: Antoine Tremblay <hexa00 at gmail dot com>
* gst/gdp/gstgdppay.c: (gst_gdp_pay_reset_streamheader):
Don't set caps on the buffers that contain a copy of the buffer
including the caps of them resulting in an always increasing refcount
of the caps and insanely large caps. Instead include a buffer without
caps in the new caps. Fixes bug #536475.
2008-06-04 05:58:38 +00:00
Sebastian Dröge
d57ab7cfdb gst/videoscale/gstvideoscale.c: Transform a given PAR to a range on the struct with the generic height/width instead ...
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
Transform a given PAR to a range on the struct with the generic
height/width instead of the struct with the possibly restricted
height/width.
2008-06-04 05:44:06 +00:00
Sebastian Dröge
8b14d08115 gst/videoscale/gstvideoscale.c: Prefer the given format if it contains something stricter than [1,MAX] for height or ...
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
Prefer the given format if it contains something stricter than [1,MAX]
for height or width and only put a structure that requires rescaling
as second. This makes it possible to use videoscale in pipelines where
the source can actually produce the wanted height/width but usually
selects a different one from the requested.
2008-06-04 04:24:27 +00:00
Sebastian Dröge
1d37b272ce gst/adder/gstadder.c: When using gst_element_iterate_pads() one has to unref every pad after usage.
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_query_duration),
(gst_adder_query_latency):
When using gst_element_iterate_pads() one has to unref every pad
after usage.
2008-06-02 12:20:35 +00:00
Sebastian Dröge
fdd708c418 gst/audioconvert/gstaudioconvert.c: Allow up to 11 positioned channels now that audioconvert can handle this but add ...
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(structure_has_fixed_channel_positions),
(gst_audio_convert_transform_caps):
Allow up to 11 positioned channels now that audioconvert can handle
this but add no default positions for > 8 channels.
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Add some unit tests for the above change: Test conversion of
11 positioned channels to stereo and the other way around, test
conversion of 15 unpositioned channels in different ways.
2008-05-30 08:42:17 +00:00
Sebastian Dröge
b86a5d4303 gst/audioconvert/gstchannelmix.c: If mixing left or right to center (or the other way around) only take the complete ...
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c:
(gst_channel_mix_fill_one_other):
If mixing left or right to center (or the other way around) only take
the complete value if we don't already have the original position in
the source.
2008-05-29 12:17:16 +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
Mark Nauwelaerts
17b17a566f gst/videorate/gstvideorate.*: React (more) to NEWSEGMENT
Original commit message from CVS:
Patch by: Mark Nauwelaerts <manauw at skynet be>
* gst/videorate/gstvideorate.c: (gst_video_rate_reset),
(gst_video_rate_flush_prev), (gst_video_rate_event),
(gst_video_rate_chain):
* gst/videorate/gstvideorate.h:
React (more) to NEWSEGMENT
Small adjustment in timestamp calculation to prevent mismatches
Fixes #435633.
2008-05-28 14:49:24 +00:00
Sebastian Dröge
57c3aa9b66 gst/adder/gstadder.c: Implement latency query.
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_query_duration),
(gst_adder_query_latency), (gst_adder_query):
Implement latency query.
2008-05-28 08:14:47 +00:00
Sebastian Dröge
4ccac97b40 gst/adder/gstadder.c: Correctly resync the iterator if gst_iterator_next() returns
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_query_duration):
Correctly resync the iterator if gst_iterator_next() returns
GST_ITERATOR_RESYNC.
2008-05-27 18:10:00 +00:00
Wim Taymans
514b8fa456 gst/playback/gstdecodebin2.c: Check for NULL cases and log them, creating ghostpads can, for example, fail when the p...
Original commit message from CVS:
* gst/playback/gstdecodebin2.c:
(gst_decode_group_control_source_pad), (gst_decode_group_expose):
Check for NULL cases and log them, creating ghostpads can, for example,
fail when the pad returns wrong caps.
* gst/playback/gstplaybin2.c: (perform_eos):
When pushing out the EOS event, collect the return value and warn when
something failed.
2008-05-27 10:35:55 +00:00
Tim-Philipp Müller
fa38b99379 gst/videoscale/gstvideoscale.c: Change default scaling method from nearest-neighbour to bilinear.
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c: (DEFAULT_PROP_METHOD):
Change default scaling method from nearest-neighbour to bilinear.
2008-05-26 10:29:20 +00:00
Tim-Philipp Müller
206f91995b Limit duration to a maximum of five seconds for tmplayer format where we can guess the duration only from the timesta...
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (parser_state_init),
(gst_sub_parse_format_autodetect), (handle_buffer):
* gst/subparse/gstsubparse.h:
* tests/check/elements/subparse.c: (test_tmplayer_style3b):
Limit duration to a maximum of five seconds for tmplayer format where
we can guess the duration only from the timestamp of the next line of
text. We don't want to show a text for eternities just because nothing
else is being said for a while.
2008-05-25 20:51:35 +00:00
Tim-Philipp Müller
747d52adb3 gst/videotestsrc/videotestsrc.c: May just as well use the precalculated uvstride here.
Original commit message from CVS:
* gst/videotestsrc/videotestsrc.c: (paint_hline_NV12_NV21):
May just as well use the precalculated uvstride here.
2008-05-22 22:35:40 +00:00
Jan Schmidt
d58def621b Add some documentation comments, and some new headers to be scanned.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-overrides.txt:
* 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/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-queue2.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* ext/cdparanoia/gstcdparanoiasrc.c:
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggdemux.h:
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggmux.h:
* gst/audioconvert/audioconvert.c:
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioconvert.h:
* gst/gdp/gstgdpdepay.h:
* gst/gdp/gstgdppay.h:
* gst/playback/gstdecodebin.c:
* gst/playback/gstdecodebin2.c:
* gst/playback/gstplaybin.c:
* gst/playback/gstplaybin2.c:
* gst/playback/gsturidecodebin.c:
* gst/tcp/gstmultifdsink.c:
* gst/tcp/gstmultifdsink.h:
* gst/tcp/gsttcp.h:
Add some documentation comments, and some new headers to be scanned.
Rename some internal enum declarations (audioconvert's DitherType and
NoiseShapingType, GstUnitType from the TCP elements) to match the
documented GObject type names so that the docs pick them up.
Name the playbin2 docs markups properly so they get picked up. They'll
need renaming back when/if playbin2 becomes playbin.
100% symbol coverage for the plugin docs, booya.
2008-05-22 22:09:16 +00:00
Thijs Vermeir
88b1e8efcf gst/videotestsrc/videotestsrc.c: Fix generation of NV12/NV21 frames. Fixes bug #532454.
Original commit message from CVS:
Patch by: Thijs Vermeir <thijsvermeir@gmail.com>
* gst/videotestsrc/videotestsrc.c: (paint_hline_NV12_NV21):
Fix generation of NV12/NV21 frames. Fixes bug #532454.
2008-05-22 18:30:15 +00:00
Sjoerd Simons
1c424d9d93 gst/playback/gstdecodebin.c: Lock the fakesink before setting the state to NULL and removing it from the bin so that ...
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/playback/gstdecodebin.c: (remove_fakesink):
Lock the fakesink before setting the state to NULL and removing it from
the bin so that a concurrent state change cannot interfere.
Fixes #534331.
2008-05-22 11:59:33 +00:00
Julien Moutte
0f80e462d9 gst/tcp/gstmultifdsink.c: Use IPPROTO_IP instead of SOL_IP, works on more platforms.
Original commit message from CVS:
2008-05-21  Julien Moutte  <julien@fluendo.com>

* gst/tcp/gstmultifdsink.c: (setup_dscp_client): Use IPPROTO_IP
instead of SOL_IP, works on more platforms.
* gst/typefind/gsttypefindfunctions.c: (aac_type_find): Fix printf
arguments.
2008-05-21 16:47:58 +00:00
Wim Taymans
2cdf18edff Some debug and comment fixes.
Original commit message from CVS:
* ext/vorbis/vorbisdec.c:
* gst/videoscale/gstvideoscale.c: (gst_video_scale_transform):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_show_frame):
Some debug and comment fixes.
* tests/examples/dynamic/addstream.c: (main):
Fix , to ;
2008-05-21 16:44:15 +00:00
Wim Taymans
c6b54c3d02 Don't use bad gst_element_get_pad().
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind):
* gst/playback/decodetest.c: (new_decoded_pad_cb):
* gst/playback/gstdecodebin.c: (gst_decode_bin_init),
(try_to_link_1), (elem_is_dynamic), (close_link), (type_found),
(cleanup_decodebin):
* gst/playback/gstdecodebin2.c: (gst_decode_bin_init),
(connect_element), (gst_decode_group_control_demuxer_pad):
* gst/playback/gstplaybasebin.c: (queue_remove_probe),
(queue_out_of_data), (gen_preroll_element), (preroll_unlinked),
(mute_group_type):
* gst/playback/gstplaybin.c: (gst_play_bin_vis_blocked),
(gst_play_bin_set_property), (handoff), (gen_video_element),
(gen_text_element), (gen_audio_element), (gen_vis_element),
(remove_sinks), (add_sink), (setup_sinks):
* gst/playback/gstplaybin2.c: (pad_added_cb), (no_more_pads_cb):
* gst/playback/gstplaysink.c: (gst_play_sink_get_video_sink),
(gst_play_sink_get_audio_sink), (gst_play_sink_vis_unblocked),
(gst_play_sink_vis_blocked), (gst_play_sink_set_vis_plugin),
(gst_play_sink_get_vis_plugin), (gst_play_sink_set_mute),
(gen_video_chain), (gen_text_chain), (gen_audio_chain),
(gen_vis_chain), (gst_play_sink_reconfigure),
(gst_play_sink_set_font_desc), (gst_play_sink_get_font_desc),
(gst_play_sink_request_pad):
* gst/playback/gsturidecodebin.c: (type_found), (setup_source):
* gst/playback/test.c: (gen_video_element), (gen_audio_element),
(cb_newpad):
* gst/playback/test6.c: (new_decoded_pad_cb):
* tests/check/elements/audioconvert.c: (GST_START_TEST):
* tests/check/elements/audiorate.c: (test_injector_chain),
(do_perfect_stream_test):
* tests/check/elements/ffmpegcolorspace.c: (GST_START_TEST):
* tests/check/elements/gdpdepay.c: (GST_START_TEST):
* tests/check/elements/gnomevfssink.c:
* tests/check/elements/textoverlay.c:
(notgst_check_setup_src_pad2), (notgst_check_teardown_src_pad2):
* tests/check/elements/videotestsrc.c: (GST_START_TEST):
* tests/check/libs/cddabasesrc.c: (GST_START_TEST):
* tests/check/pipelines/oggmux.c: (test_pipeline):
* tests/check/pipelines/streamheader.c: (GST_START_TEST):
* tests/check/pipelines/theoraenc.c: (GST_START_TEST):
* tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
* tests/examples/seek/scrubby.c: (make_wav_pipeline):
* tests/examples/seek/seek.c: (make_mod_pipeline),
(make_dv_pipeline), (make_wav_pipeline), (make_flac_pipeline),
(make_sid_pipeline), (make_parse_pipeline), (make_vorbis_pipeline),
(make_theora_pipeline), (make_vorbis_theora_pipeline),
(make_avi_msmpeg4v3_mp3_pipeline), (make_mp3_pipeline),
(make_avi_pipeline), (make_mpeg_pipeline), (make_mpegnt_pipeline),
(update_fill), (msg_buffering):
Don't use bad gst_element_get_pad().
2008-05-21 16:36:50 +00:00
Sebastian Dröge
736b181916 gst/audioconvert/gstchannelmix.c: Fix copy & paste error in last commit.
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
Fix copy & paste error in last commit.
2008-05-21 11:36:37 +00:00
Sebastian Dröge
7d605d4514 gst/audioconvert/gstchannelmix.c: Add support for mixing GST_AUDIO_CHANNEL_POSITION_SIDE_* from/to other channel posi...
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
Add support for mixing GST_AUDIO_CHANNEL_POSITION_SIDE_* from/to
other channel positions when source has SIDE channels and dest doesn't
or the other way around.
2008-05-21 11:30:58 +00:00
Henrik Eriksson
10ae17ced1 gst/tcp/gstmultifdsink.*: Add support for DSCP QOS. Fixes #469933.
Original commit message from CVS:
Patch by: Henrik Eriksson <henriken at axis dot com>
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
(gst_multi_fd_sink_init), (setup_dscp_client), (setup_dscp),
(gst_multi_fd_sink_add_full), (gst_multi_fd_sink_set_property),
(gst_multi_fd_sink_get_property):
* gst/tcp/gstmultifdsink.h:
Add support for DSCP QOS. Fixes #469933.
2008-05-21 11:29:25 +00:00
Sebastian Dröge
d47bd6d7bc gst/audioconvert/gstchannelmix.c: Prevent division by zero if the channel mix matrix contains only zeroes.
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c:
(gst_channel_mix_fill_normalize):
Prevent division by zero if the channel mix matrix contains only
zeroes.
2008-05-21 07:28:04 +00:00
Antoine Tremblay
a8dda35c1b gst/gdp/gstgdppay.c: Close a buffer memory leak. Fixes bug #534071.
Original commit message from CVS:
Patch by: Antoine Tremblay <hexa00 at gmail dot com>
* gst/gdp/gstgdppay.c: (gst_gdp_pay_chain):
Close a buffer memory leak. Fixes bug #534071.
2008-05-21 06:45:22 +00:00
Sebastian Dröge
e66b0a6642 gst/adder/gstadder.c: Adder also doesn't support audio/x-raw-int with width!=depth so don't claim this on the pad tem...
Original commit message from CVS:
* gst/adder/gstadder.c:
Adder also doesn't support audio/x-raw-int with width!=depth so don't
claim this on the pad template caps.
2008-05-21 05:48:05 +00:00
Sebastian Dröge
fcda3964dc gst/audioconvert/gstaudioconvert.c: Fix logic in last commit.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_fixate_channels):
Fix logic in last commit.
2008-05-20 12:26:32 +00:00
Sebastian Dröge
d76c4b4c65 gst/audioconvert/gstaudioconvert.c: Passthrough the channel positions if the number of output channels is the same as...
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_fixate_channels):
Passthrough the channel positions if the number of output channels is
the same as the number of input channels, the input had a channel
layout and downstream requests no special one. We did this already for
> 2 channels but now it's also done for 1 channel. Fixes bug #533617.
2008-05-20 12:15:34 +00:00
Sebastian Dröge
b5a5d64713 gst/audioconvert/gstaudioconvert.c: Correctly set the default channel positions when converting to 8 channels.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_fixate_channels):
Correctly set the default channel positions when converting to 8
channels.
2008-05-20 08:12:19 +00:00
Tim-Philipp Müller
7cb1276dac gst/typefind/gsttypefindfunctions.c: Use data scan helper in aac typefinder and stop scanning for headers when we've ...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (aac_type_find):
Use data scan helper in aac typefinder and stop scanning
for headers when we've found a type. Also fix potential invalid
memory access when calculating the frame length.
2008-05-19 15:59:40 +00:00
Tim-Philipp Müller
cfc8f3c0d7 gst/typefind/gsttypefindfunctions.c: Don't modify scan context when we return FALSE in ensure_data, so it's possible ...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (data_scan_ctx_ensure_data),
(mpeg_sys_is_valid_pack):
Don't modify scan context when we return FALSE in ensure_data, so
it's possible to continue scanning, and we don't end up with a NULL
data pointer and a positive size, which might bite us the next time
we're called. Small constification.
2008-05-19 14:09:08 +00:00
Sebastian Dröge
05cf63634e gst/adder/gstadder.c: Adder doesn't support 24 bit samples so don't claim it supports them in the pad template caps.
Original commit message from CVS:
* gst/adder/gstadder.c:
Adder doesn't support 24 bit samples so don't claim it supports them
in the pad template caps.
2008-05-16 21:12:02 +00:00
Tim-Philipp Müller
d92ff26d29 gst/audioresample/gstaudioresample.c: Revert previous change which made basetransform handle buffer_alloc and which b...
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
Revert previous change which made basetransform handle buffer_alloc
and which breaks things badly in the non-passthrough case since it
returned buffers with a different (ie. sometimes smaller) size than
the size requested.
2008-05-14 13:57:41 +00:00
Sebastian Dröge
6720c5beb8 gst/adder/gstadder.c: Correctly declare the supported endianness on the pad templates and check for correct endiannes...
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_setcaps),
(gst_adder_class_init):
Correctly declare the supported endianness on the pad templates
and check for correct endianness in the set caps function. Adder
only supports native endianness.
Also use gst_element_class_set_details_simple().
2008-05-14 10:58:52 +00:00
Hannes Bistry
b9bc12afd8 gst/tcp/: Fix regression in clientsrc because we did not add the fd to the poll set anymore. Fixes #532364.
Original commit message from CVS:
Patch by: Hannes Bistry <hannesb at gmx dot de>
* gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_start):
* gst/tcp/gsttcpserversink.c:
(gst_tcp_server_sink_handle_server_read),
(gst_tcp_server_sink_handle_wait), (gst_tcp_server_sink_init_send):
Fix regression in clientsrc because we did not add the fd to the poll
set anymore. Fixes #532364.
Do some cleanups here and there.
2008-05-13 16:02:19 +00:00
Sebastian Dröge
05349cc354 gst/playback/: Use correct marshallers. GstCaps are a boxed type and no GObject subclass.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
* gst/playback/gstplay-marshal.list:
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init):
Use correct marshallers. GstCaps are a boxed type and no GObject
subclass.
2008-05-13 13:04:24 +00:00
Sebastian Dröge
4d5870847f gst/ffmpegcolorspace/imgconvert.c: Fix nv12<->nv21 conversion if stride is larger than width.
Original commit message from CVS:
* gst/ffmpegcolorspace/imgconvert.c: (nv12_to_nv21):
Fix nv12<->nv21 conversion if stride is larger than width.
2008-05-13 09:14:44 +00:00
Tim-Philipp Müller
fed34307db gst/typefind/gsttypefindfunctions.c: Don't do lots of 4-byte peeks, but use the 'new' data scan helper for this inste...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (h264_video_type_find):
Don't do lots of 4-byte peeks, but use the 'new' data scan helper
for this instead; don't check if we've found enough markers after
each and every step, it's enough to do that only if we've actually
found a new marker.
Embed a G_UNLIKELY into the IS_MPEG_HEADER macro.
2008-05-10 20:16:21 +00:00
Tim-Philipp Müller
104fed4d66 gst/typefind/gsttypefindfunctions.c: Move scan helper thingy to the beginning of the file so we can use it in other t...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c:
(DATA_SCAN_CTX_CHUNK_SIZE), (DataScanCtx), (data_scan_ctx_advance),
(data_scan_ctx_ensure_data), (GST_MPEGVID_TYPEFIND_TRY_SYNC),
(mpeg_video_stream_type_find):
Move scan helper thingy to the beginning of the file so we can use
it in other typefind functions. Rename it to something more
generic. Also improve handling of things towards the end of the
typefind data: peek as much as we can if we know the size of the
data, rather than just min_size.
2008-05-10 18:19:17 +00:00
Sebastian Dröge
531c6fb462 gst/ffmpegcolorspace/: Add conversions from/to NV12 and NV21 and conversions between those two formats. Fixes bug #53...
Original commit message from CVS:
Based on a patch by:
Björn Benderius <bjoern dot benderius at axis dot com>
* 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: (nv12_to_nv21):
* gst/ffmpegcolorspace/imgconvert_template.h:
Add conversions from/to NV12 and NV21 and conversions between those
two formats. Fixes bug #532166.
2008-05-09 08:34:52 +00:00
Edward Hervey
9fa3d7a294 gst/typefind/gsttypefindfunctions.c: Abort the h264 typefinding as soon as _peek() doesn't return anything, which hap...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (h264_video_type_find):
Abort the h264 typefinding as soon as _peek() doesn't return anything,
which happens for example with files smaller than 128kb.
2008-05-08 17:35:44 +00:00
Sjoerd Simons
09163ca363 gst/audioresample/gstaudioresample.c: Let audioresample use the buffer allocation of basetransform instead of it's ow...
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
Let audioresample use the buffer allocation of basetransform instead
of it's own stuff.
* tests/check/elements/audioresample.c: (alloc_only_48000),
(GST_START_TEST), (audioresample_suite):
Add unit test for the recent basetransform bugfix, where upstream
changes caps to something that can't be passed through anymore.
2008-05-08 06:20:42 +00:00
Sebastian Dröge
b9a285021c gst/volume/gstvolume.c: Return NOT_NEGOTIATED if we didn't set a process function yet for some reason instead of cras...
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_transform_ip):
Return NOT_NEGOTIATED if we didn't set a process function yet for some
reason instead of crashing later. Might fix bug #509125.
2008-05-06 12:35:09 +00:00
Tim-Philipp Müller
fd54092a2a gst/audioconvert/: Add support for more than 8 channels and NONE channel layouts. For more than 8 channels no channel...
Original commit message from CVS:
Based on a patch by: Tim-Philipp Müller  <tim.muller at collabora co uk>
* gst/audioconvert/audioconvert.c: (audio_convert_prepare_context):
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_parse_caps),
(structure_has_fixed_channel_positions),
(gst_audio_convert_transform_caps):
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_matrix):
Add support for more than 8 channels and NONE channel layouts. For
more than 8 channels no channel conversion is supported yet, only
format conversions are supported. Fixes bug #398033.
* tests/check/elements/audioconvert.c: (verify_convert),
(GST_START_TEST), (audioconvert_suite):
Add some unit tests by Tim for checking the NONE channel layouts
and more than 8 channels and add some more unit tests for channel
conversions.
2008-05-06 12:12:16 +00:00
Wim Taymans
4a3db41f6d gst/playback/gstdecodebin2.c: When autoplugging fails, set the element back to NULL before unreffing it.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (connect_pad):
When autoplugging fails, set the element back to NULL before
unreffing it.
2008-05-06 10:16:49 +00:00
Sebastian Dröge
9333eb4899 gst/subparse/samiparse.c: Remove trailing, leading and double whitespaces.
Original commit message from CVS:
* gst/subparse/samiparse.c: (handle_start_sync),
(end_sami_element), (characters_sami):
Remove trailing, leading and double whitespaces.
Correctly timestamp buffers and output the last buffer too.
* tests/check/elements/subparse.c: (GST_START_TEST),
(subparse_suite):
Add a simple unit test for SAMI parsing.
2008-05-05 12:33:05 +00:00
Young-Ho Cha
76e3ffb61c gst/subparse/samiparse.c: Only output characters inside the "sync" elements. There could be other elements like "styl...
Original commit message from CVS:
Patch by: Young-Ho Cha <ganadist at chollian dot net>
* gst/subparse/samiparse.c: (handle_start_sync),
(start_sami_element), (end_sami_element), (characters_sami),
(sami_context_reset):
Only output characters inside the "sync" elements. There could be
other elements like "style" that have some content but should
not be printed. Fixes bug #467911.
2008-05-05 11:14:48 +00:00
Sebastian Dröge
de277a5b2a gst/playback/: Allow setting -1 as current-audio to mute the current audio stream, similar to what is done for subtit...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (set_audio_mute),
(set_active_source):
* gst/playback/gstplaybasebin.h:
* gst/playback/gstplaybin.c: (gst_play_bin_class_init),
(playbin_set_audio_mute):
Allow setting -1 as current-audio to mute the current audio stream,
similar to what is done for subtitles. Fixes bug #342294.
2008-05-05 10:03:51 +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
Tim-Philipp Müller
005c1c8636 gst/subparse/: Fix parsing of tmplayer subtitle variant where every single line contains text and there isn't an empt...
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (handle_buffer),
(gst_sub_parse_sink_event):
* gst/subparse/tmplayerparse.c: (tmplayer_process_buffer),
(tmplayer_parse_line):
Fix parsing of tmplayer subtitle variant where every single line contains
text and there isn't an empty line after each line to determine the
duration (#530962). Improve EOS handling for tmplayer subtitles a bit by
making sure that we push out the last line of text without a duration if
there's still text left in the buffer at the end.
2008-05-03 15:45:23 +00:00
Tim-Philipp Müller
ee90cf1969 gst/subparse/gstsubparse.c: Fix detection of discontinuities based on the buffer offset (doesn't work so well if no b...
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (feed_textbuf):
Fix detection of discontinuities based on the buffer offset (doesn't work
so well if no buffer offset is set) and also check for the DISCONT buffer
flag. This keeps the parser state from being reset after each buffer in
the unit test.
2008-05-03 15:39:04 +00:00
Tim-Philipp Müller
6de5983831 gst/typefind/gsttypefindfunctions.c: Further fine-tuning: don't absolutely require sequence or GOP headers but adjust...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mpeg_video_stream_type_find):
Further fine-tuning: don't absolutely require sequence or GOP headers
(as introduced in the previous commit), but adjust the typefind
probabilities returned accordingly if we don't see them. Also make sure
picture header and first slice are somewhat close to each other (which
is not perfect but still better than requiring a fixed offset or having
no limit at all).
2008-05-03 12:09:16 +00:00
Stefan Kost
2b843ca69f gst/playback/: Also include config.h when relying on defines from it. Fixes the build. Its been a please to serve :)
Original commit message from CVS:
* gst/playback/test4.c:
* gst/playback/test5.c:
* gst/playback/test6.c:
* gst/playback/test7.c:
Also include config.h when relying on defines from it. Fixes the
build. Its been a please to serve :)
2008-05-02 11:13:05 +00:00
Thijs Vermeir
32c304ad6f Add support for NV12 and NV21 in videotestsrc
Original commit message from CVS:
* gst/videotestsrc/videotestsrc.c (paint_setup_NV12),
(paint_setup_NV21), (paint_hline_NV12_NV21):
Add support for NV12 and NV21 in videotestsrc
2008-05-02 10:54:51 +00:00
Sebastian Dröge
abbce230e2 gst/videoscale/: Support 1x1 images as input and output as for example the BBC HQ new streams have 1x1 GIFs in the pl...
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c:
* gst/videoscale/vs_4tap.c: (vs_image_scale_4tap_Y):
* gst/videoscale/vs_image.c: (vs_image_scale_nearest_RGBA),
(vs_image_scale_linear_RGBA), (vs_image_scale_nearest_RGB),
(vs_image_scale_linear_RGB), (vs_image_scale_nearest_YUYV),
(vs_image_scale_linear_YUYV), (vs_image_scale_nearest_UYVY),
(vs_image_scale_linear_UYVY), (vs_image_scale_nearest_Y),
(vs_image_scale_linear_Y), (vs_image_scale_nearest_RGB565),
(vs_image_scale_linear_RGB565), (vs_image_scale_nearest_RGB555),
(vs_image_scale_linear_RGB555):
Support 1x1 images as input and output as for example the BBC HQ new
streams have 1x1 GIFs in the playlists for some reason.
2008-05-02 10:02:05 +00:00
Tim-Philipp Müller
ea0d78e8e5 gst/playback/gstdecodebin.c: If we can't activate one of the decoders we plugged in (such as, say, musepackdec) for s...
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (free_pad_probe_for_element),
(try_to_link_1):
If we can't activate one of the decoders we plugged in (such as,
say, musepackdec) for some reason (it might not support push mode,
for example), remove any pad probes that close_pad_link() might
have set up. This makes sure we later don't try to remove a probe
for a pad that doesn't exist any longer, and avoids nast warnings
and probably other things too.
2008-05-01 19:11:42 +00:00
Tim-Philipp Müller
f8977b9e9e gst/typefind/gsttypefindfunctions.c: Rework mpeg video stream typefinding a bit more: make sure sequence,
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c:
(mpeg_video_stream_ctx_ensure_data), (mpeg_video_stream_type_find),
(plugin_init):
Rework mpeg video stream typefinding a bit more: make sure sequence,
GOP, picture and slice headers appear in the order they should and
that we've in fact at least had one of each; fix picture header
detection; decouple picture and slice header check - don't assume
they're at a fixed offset, there may be extra data in between. Also,
announce varying degrees of probability depending on what we found
exactly (multiple pictures, at least one picture, just sequence and
GOP headers). Finally, in _ensure_data(), take into account that we
might be typefinding smaller amounts of data, such as the first
buffer of a stream, so fall back to the minimum size needed as long
as that's available, instead of erroring out if there's less than
2kB of data. Fixes #526173. Conveniently also doesn't recognise the
fuzzed file from #399342 as valid.
2008-04-30 20:54:56 +00:00
Tim-Philipp Müller
5f6db60a4d gst/typefind/gsttypefindfunctions.c: Refactor a bit: use context structure to track parsing offset and size of availa...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (MpegVideoStreamCtx),
(mpeg_video_stream_ctx_advance), (mpeg_video_stream_ctx_ensure_data),
(mpeg_video_stream_type_find):
Refactor a bit: use context structure to track parsing offset and size of
available data and make the code a bit clearer. Fixes bad memory access
in #356937.
2008-04-30 14:37:52 +00:00
Michael Smith
802c45b10b gst/: Include stdlib.h and unistd.h only if the appropriate HAVE_*_H macro is defined.
Original commit message from CVS:
* gst/playback/test4.c:
* gst/playback/test5.c:
* gst/playback/test6.c:
* gst/tcp/gstmultifdsink.c:
Include stdlib.h and unistd.h only if the appropriate HAVE_*_H macro
is defined.
2008-04-28 22:18:49 +00:00
Stefan Kost
2b44c294ff gst/playback/gstplaybin.c: Remove obsolete streaminfo code and fix a leak. Fixes #529546
Original commit message from CVS:
* gst/playback/gstplaybin.c:
Remove obsolete streaminfo code and fix a leak. Fixes #529546
2008-04-24 08:19:35 +00:00
Sebastian Dröge
0c73cdcbc8 gst/typefind/gsttypefindfunctions.c: Add "mpp" and "mp+" as possible extensions for MusePack files.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (musepack_type_find),
(plugin_init):
Add "mpp" and "mp+" as possible extensions for MusePack files.
Add typefinding for MusePack StreamVersion 8 files and include the
stream version in the caps.
2008-04-19 20:06:59 +00:00
Edward Hervey
9e630c23db gst/gdp/gstgdpdepay.c: Don't validate the payload if there isn't any.
Original commit message from CVS:
* gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
Don't validate the payload if there isn't any.
Fixes #525915
2008-04-18 14:54:01 +00:00
Stefan Kost
e6528c39fe gst/audiotestsrc/gstaudiotestsrc.c: Remove cpp style commented old code.
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
Remove cpp style commented old code.
2008-04-15 19:06:00 +00:00
Stefan Kost
0d3c154ccf gst/playback/gstdecodebin2.c: Fix signal docs.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c:
Fix signal docs.
2008-04-15 19:02:10 +00:00
Wim Taymans
f0738f6fd3 docs/design/draft-keyframe-force.txt: Fix typo.
Original commit message from CVS:
* docs/design/draft-keyframe-force.txt:
Fix typo.
* gst/playback/gstqueue2.c: (update_buffering),
(gst_queue_handle_src_query):
Set buffering mode in the messages.
Set buffering percent in the query.
* tests/examples/seek/seek.c: (update_fill), (msg_state_changed),
(do_stream_buffering), (do_download_buffering), (msg_buffering):
Do some more fancy things based on the buffering method in use.
2008-04-11 01:25:01 +00:00
Wim Taymans
e5bdd95038 gst/playback/gstqueue2.c: Include extra buffering stats in the buffering message.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (update_buffering),
(gst_queue_close_temp_location_file), (gst_queue_handle_src_query),
(gst_queue_src_checkgetrange_function):
Include extra buffering stats in the buffering message.
Implement BUFFERING query.
* gst/playback/gsturidecodebin.c: (do_async_start),
(do_async_done), (type_found), (setup_streaming), (setup_source),
(gst_uri_decode_bin_change_state):
Only add decodebin2 when the type is found in streaming mode.
Make uridecodebin async to PAUSED even when we don't have decodebin2
added yet.
2008-04-09 21:40:17 +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
Stefan Kost
b04f8ef354 docs/plugins/: Add playbin, playbin2, decodebin, decodebin2, uridecodebin and oggmux
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-overrides.txt:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
Add playbin, playbin2, decodebin, decodebin2, uridecodebin and oggmux
* 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/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-queue2.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
Update introspection data.
* ext/ogg/gstoggmux.c:
Document oggmux.
* gst/playback/gstdecodebin2.c:
Don't use gtk-doc style comment start for private stuff, but make it
formatted like this for consistency.
2008-04-03 14:58:06 +00:00
Wim Taymans
c98a370f8c gst/playback/gstdecodebin2.c: Remove fakesink hack, we can now implement this more elegantly.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
(gst_decode_bin_init), (gst_decode_bin_dispose),
(gst_decode_bin_set_sink_caps), (gst_decode_bin_get_sink_caps),
(gst_decode_bin_set_property), (gst_decode_bin_get_property),
(analyze_new_pad), (connect_pad), (expose_pad),
(gst_decode_group_new), (gst_decode_group_control_demuxer_pad),
(gst_decode_group_expose), (gst_decode_group_free),
(do_async_start), (do_async_done), (gst_decode_bin_change_state):
Remove fakesink hack, we can now implement this more elegantly.
Added property to bypass typefinding.
Removed underrun callback and demuxer pad probe, we now use the srcpad
probe to expose groups.
API::sink-caps property
* gst/playback/gstplaybin2.c: (no_more_pads_cb):
Guard against multiple emissions of the no_more_pads signal, which
happens when we are dealing with chained oggs.
* gst/playback/gsturidecodebin.c: (remove_decoders),
(make_decoder), (type_found), (setup_streaming), (source_new_pad),
(setup_source):
For streams, use our own typefind element and plug our queue after it.
We will need this to determine the type of buffering to use for the
queue soon.
2008-04-03 12:16:04 +00:00
Wim Taymans
8f77a5d928 gst/playback/gstqueue2.c: Update the estimated input data when we push out a buffer.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (update_out_rates),
(gst_queue_open_temp_location_file),
(gst_queue_close_temp_location_file), (gst_queue_handle_src_event),
(gst_queue_handle_src_query), (gst_queue_set_property):
Update the estimated input data when we push out a buffer.
Add some debug info about the temp file.
Only forward src events when we are not using a temp file.
Don't block the duration query, we need to find something better.
Don't leak the temp filename.
2008-04-02 11:08:05 +00:00
Sebastian Dröge
c82eca96a6 configure.ac: Require GLib 2.12 and liboil 0.3.14.
Original commit message from CVS:
* configure.ac:
Require GLib 2.12 and liboil 0.3.14.
* gst/volume/gstvolume.c: (volume_process_double):
Unconditionally use liboil 0.3.14 function.
2008-04-01 14:01:14 +00:00
Michael Smith
670d6cd1e4 gst/gdp/gstgdpdepay.c: Check the body CRC (if set) when depayloading.
Original commit message from CVS:
* gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
Check the body CRC (if set) when depayloading.
Fixes #522401.
2008-03-27 15:26:38 +00:00
Wim Taymans
03e571d945 gst/playback/gstqueue2.c: The queue is never filled when there are no buffers in the queue at all.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (gst_queue_is_filled):
The queue is never filled when there are no buffers in the queue at all.
Fixes #523993.
2008-03-24 14:08:22 +00:00
Wim Taymans
ad1cbe1edd gst/playback/gstplaybin2.c: Update some docs.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
(init_group), (free_group), (gst_play_bin_init),
(gst_play_bin_finalize), (gst_play_bin_set_uri),
(gst_play_bin_set_suburi), (gst_play_bin_get_video_tags),
(gst_play_bin_get_audio_tags), (gst_play_bin_get_text_tags),
(gst_play_bin_set_current_video_stream),
(gst_play_bin_set_current_audio_stream),
(gst_play_bin_set_current_text_stream),
(gst_play_bin_set_encoding), (gst_play_bin_set_property),
(gst_play_bin_get_property), (pad_added_cb), (pad_removed_cb),
(no_more_pads_cb), (perform_eos), (autoplug_select_cb),
(activate_group), (deactivate_group), (setup_next_source),
(save_current_group), (gst_play_bin_change_state):
Update some docs.
Add new locks and conds to protect pipeline creation and group
switching.
Implement the sub-uri property.
Keep track of pending uridecodebin creation and configure the output
pipeline after all streams are configured.
Propagate subtitle encoding to the uridecodebins.
Implement getting the video/audio/visualisation elements.
Use input-selector for stream switching.
If we are asked to do visualisation, prefer to autoplug raw sinks
instead of sinks that accept encoded data.
2008-03-24 12:26:30 +00:00
Wim Taymans
cd1fed3345 gst/playback/gstplaysink.*: Add methods to get audio/video/vis elements.
Original commit message from CVS:
* gst/playback/gstplaysink.c: (gst_play_sink_class_init),
(gst_play_sink_init), (gst_play_sink_dispose),
(gst_play_sink_set_video_sink), (gst_play_sink_get_video_sink),
(gst_play_sink_set_audio_sink), (gst_play_sink_get_audio_sink),
(gst_play_sink_vis_unblocked), (gst_play_sink_vis_blocked),
(gst_play_sink_set_vis_plugin), (gst_play_sink_get_vis_plugin),
(gst_play_sink_set_volume), (gst_play_sink_get_volume),
(gst_play_sink_set_mute), (gen_video_chain), (gen_text_chain),
(gen_audio_chain), (gen_vis_chain), (gst_play_sink_reconfigure),
(gst_play_sink_set_font_desc), (gst_play_sink_get_font_desc),
(gst_play_sink_send_event_to_sink), (gst_play_sink_change_state):
* gst/playback/gstplaysink.h:
Add methods to get audio/video/vis elements.
Add methods to set the font description for the overlay.
Remove properties, we're using this element with its methods only.
Add support for subtitles.
Rearrange the locking a bit to not use the object lock for protecting
the pipeline construction.
Try to use the volume and mute property on the sink when its available.
Implement the mute option with volume when the sink does not have a mute
property.
Only add volume element when the sink has no volume property.
Only do visualisations with raw audio pads.
2008-03-24 12:15:26 +00:00
Wim Taymans
3640ae2d36 gst/playback/gsturidecodebin.c: Add a readonly source property and notify.
Original commit message from CVS:
* gst/playback/gsturidecodebin.c:
(gst_uri_decode_bin_autoplug_factories),
(gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
(gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_encoding),
(gst_uri_decode_bin_set_property),
(gst_uri_decode_bin_get_property), (no_more_pads_full),
(new_decoded_pad_cb), (gen_source_element), (remove_decoders),
(proxy_autoplug_factories_signal), (make_decoder),
(source_new_pad), (setup_source):
Add a readonly source property and notify.
Add new lock for protecting the construction of the pipeline.
Keep track of the decodebins we plugged.
Correctly proxy the autoplug signal so that it actually continues.
Proxy subtitle-encoding to the decodebins.
2008-03-24 11:54:02 +00:00
Wim Taymans
8eb84372bd gst/playback/gstdecodebin2.c: Protect caps property with the object lock.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_dispose),
(gst_decode_bin_set_caps), (gst_decode_bin_get_caps),
(gst_decode_bin_set_subs_encoding),
(gst_decode_bin_get_subs_encoding),
(gst_decode_bin_autoplug_factories), (connect_pad), (are_raw_caps),
(deactivate_free_recursive):
Protect caps property with the object lock.
Protect encoding property with the object lock.
Keep list of elements we added that have the subtitle-encoding property.
Distribute the subtitle-encoding to all of the elements when it
changes.
2008-03-24 11:36:08 +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
Sebastian Dröge
2034387d4d gst/volume/gstvolume.c: memset buffers to zero if we get a GAP buffer. We usually see a buffer as one unit so let's h...
Original commit message from CVS:
* gst/volume/gstvolume.c: (gst_volume_interface_supported),
(gst_volume_base_init), (gst_volume_class_init),
(volume_process_double), (volume_process_float),
(volume_transform_ip), (plugin_init):
memset buffers to zero if we get a GAP buffer. We usually see a
buffer as one unit so let's handle it as one and don't care about
volume changes while processing one buffer.
Also clean up some stuff a bit.
2008-03-21 16:46:33 +00:00
Sebastian Dröge
88136fc11a gst/audioconvert/gstaudioconvert.c: Make audioconvert GAP-aware by outputting silence buffers when the input has the ...
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init),
(gst_audio_convert_create_silence_buffer),
(gst_audio_convert_transform):
Make audioconvert GAP-aware by outputting silence buffers when the
input has the GAP flag set. This is up to 8x faster.
Based on a patch by Stefan Kost. Fixes bug #517813.
2008-03-21 15:58:44 +00:00
Sebastian Dröge
cd5f49f47f gst/volume/gstvolume.c: Use oil_scalarmultiply_f64_ns() for double processing when it's available at compile time.
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_process_double):
Use oil_scalarmultiply_f64_ns() for double processing when it's
available at compile time.
2008-03-21 15:54:54 +00:00
David Schleef
66935a9872 gst/videotestsrc/videotestsrc.c: Oops, revert last change because -base is in freeze.
Original commit message from CVS:
* gst/videotestsrc/videotestsrc.c:  Oops, revert last change
because -base is in freeze.
2008-03-14 18:42:35 +00:00
William M. Brack
bfdcf07674 gst/videotestsrc/videotestsrc.c: Fix Bayer pattern generation.
Original commit message from CVS:
Patch by: William M. Brack
* gst/videotestsrc/videotestsrc.c: Fix Bayer pattern generation.
2008-03-14 17:33:09 +00:00
Wim Taymans
4803dd99e5 gst/playback/gststreamselector.*: Revert change that caused regression until a real fix is found.
Original commit message from CVS:
* gst/playback/gststreamselector.c: (gst_selector_pad_event),
(gst_selector_pad_chain):
* gst/playback/gststreamselector.h:
Revert change that caused regression until a real fix is found.
Fixes #522203.
2008-03-14 09:54:44 +00:00
Wim Taymans
6c50e0031a gst/tcp/gstmultifdsink.c: Update mode property docs, it's deprecated now.
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
Update mode property docs, it's deprecated now.
2008-03-07 16:10:51 +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
Jan Schmidt
43a120bc57 gst/Makefile.am: GST_PLUGINS_ALL correctly lists subparse and tcp now, don't distclean them twice
Original commit message from CVS:
* gst/Makefile.am:
GST_PLUGINS_ALL correctly lists subparse and tcp now, don't distclean
them twice
* win32/common/libgstinterfaces.def:
* win32/common/libgstrtp.def:
Add new API to the defs
2008-03-03 23:59:45 +00:00
Sebastian Dröge
6f86b8b8a7 gst/typefind/gsttypefindfunctions.c: Add typefinder for IMelody files, using audio/x-imelody.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
Add typefinder for IMelody files, using audio/x-imelody.
See bug #519516.
2008-03-03 06:22:39 +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
José Alburquerque
f61c20f1d2 gst/playback/gstplaybin2.c: Make the function signature of the _get_*_tags() functions match the signature of the vfu...
Original commit message from CVS:
Patch by: José Alburquerque <jaalburqu svn gnome org>
* gst/playback/gstplaybin2.c:
Make the function signature of the _get_*_tags() functions match
the signature of the vfuncs they implement, ie. return a
GstTagList rather than a GstStructure, which is more correct,
even if one is typedef'ed to the other (#518940).
2008-03-02 18:43:15 +00:00
Wim Taymans
472162d1b1 gst/playback/gstplaybin2.c: Enable vis setting.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (gst_play_bin_set_property):
Enable vis setting.
* gst/playback/gstplaysink.c: (gst_play_sink_init),
(gst_play_sink_dispose), (gst_play_sink_vis_unblocked),
(gst_play_sink_vis_blocked), (gst_play_sink_set_vis_plugin),
(gen_vis_chain):
Implement vis switching while playing.
2008-02-29 12:26:48 +00:00
Peter Kjellerstedt
405571a67e gst/tcp/: Removed fdset and stress test, they are now known as GstPoll in core.
Original commit message from CVS:
Patch by: Peter Kjellerstedt  <pkj at axis com>
* gst/tcp/Makefile.am:
* gst/tcp/fdsetstress.c:
* gst/tcp/gstfdset.c:
* gst/tcp/gstfdset.h:
Removed fdset and stress test, they are now known as GstPoll in
core.
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
(gst_multi_fd_sink_add_full), (gst_multi_fd_sink_remove),
(gst_multi_fd_sink_clear), (gst_multi_fd_sink_remove_client_link),
(gst_multi_fd_sink_handle_client_write),
(gst_multi_fd_sink_queue_buffer),
(gst_multi_fd_sink_handle_clients), (gst_multi_fd_sink_start),
(gst_multi_fd_sink_stop):
* gst/tcp/gstmultifdsink.h:
* gst/tcp/gsttcp.c: (gst_tcp_socket_read), (gst_tcp_socket_close),
(gst_tcp_read_buffer), (gst_tcp_gdp_read_buffer),
(gst_tcp_gdp_read_caps):
* gst/tcp/gsttcp.h:
* gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_init),
(gst_tcp_client_sink_setcaps), (gst_tcp_client_sink_render),
(gst_tcp_client_sink_start), (gst_tcp_client_sink_stop):
* gst/tcp/gsttcpclientsink.h:
* gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_init),
(gst_tcp_client_src_create), (gst_tcp_client_src_start),
(gst_tcp_client_src_stop), (gst_tcp_client_src_unlock):
* gst/tcp/gsttcpclientsrc.h:
* gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_handle_wait),
(gst_tcp_server_sink_init_send), (gst_tcp_server_sink_close):
* gst/tcp/gsttcpserversink.h:
* gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_init),
(gst_tcp_server_src_create), (gst_tcp_server_src_start),
(gst_tcp_server_src_stop), (gst_tcp_server_src_unlock):
* gst/tcp/gsttcpserversrc.h:
Port to GstPoll. See #505417.
2008-02-28 10:54:14 +00:00
Sebastian Dröge
929afcbaa1 gst/typefind/gsttypefindfunctions.c: Comment smoke typefinder for now. The smokedec plugin needs one frame per buffer...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
Comment smoke typefinder for now. The smokedec plugin needs one
frame per buffer but we have no parser yet, thus it simply crashes
in most situations.
2008-02-25 07:21:33 +00:00
Sebastian Dröge
9327c2a8ec gst/typefind/gsttypefindfunctions.c: Add typefinder for the smoke video codec. Copied from the jpeg plugin.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
Add typefinder for the smoke video codec. Copied from the jpeg plugin.
2008-02-25 06:48:14 +00:00
Sebastian Dröge
49e1c708bb gst/typefind/gsttypefindfunctions.c: Add midi typefinder, copied from the timidity plugin.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mid_type_find),
(plugin_init):
Add midi typefinder, copied from the timidity plugin.
2008-02-25 06:29:09 +00:00
Tomasz Sałaciński
6ab3a0e0c0 Forward slashes at the beginning and end of a line also signify italics (Fixes: #518162).
Original commit message from CVS:
Based on patch by: Tomasz Sałaciński <tsalacinski gmail com>
* gst/subparse/gstsubparse.c: (parse_mdvdsub):
* tests/check/elements/subparse.c: (test_microdvd_with_italics),
(subparse_suite):
Forward slashes at the beginning and end of a line also signify
italics (Fixes: #518162).
2008-02-23 09:51:26 +00:00
Stefan Kost
7278c5871c gst/audiotestsrc/gstaudiotestsrc.c: Mark buffers as GAP,if volume is 0.0 and fix the previous logic.
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
Mark buffers as GAP,if volume is 0.0 and fix the previous logic.
2008-02-21 08:05:10 +00:00
Wim Taymans
dd1282f45a gst/playback/gstplaysink.c: Recursively search the sink element for a last-frame property so that we can also find th...
Original commit message from CVS:
* gst/playback/gstplaysink.c: (find_property),
(gst_play_sink_find_property), (gen_video_chain),
(gst_play_sink_reconfigure), (gst_play_sink_get_last_frame):
Recursively search the sink element for a last-frame property so that we
can also find the property in autovideosink and friends that don't
always proxy the internal sink properties.
2008-02-20 11:52:28 +00:00
Josep Torra Valles
51528422ca gst/typefind/gsttypefindfunctions.c: Increase the MPEG PS typefind lookup to fix typefinding on HD clips.
Original commit message from CVS:
2008-02-19  Julien Moutte  <julien@fluendo.com>

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

* gst/typefind/gsttypefindfunctions.c: Increase the MPEG PS
typefind lookup to fix typefinding on HD clips.
2008-02-19 16:16:55 +00:00
Tim-Philipp Müller
943d4cdc35 gst/playback/gstscreenshot.*: Fix up copyright (I rewrote the GStreamer-0.10 code for this from scratch back in the d...
Original commit message from CVS:
* gst/playback/gstscreenshot.c:
* gst/playback/gstscreenshot.h:
Fix up copyright (I rewrote the GStreamer-0.10 code for
this from scratch back in the days).
2008-02-19 15:50:37 +00:00
Wim Taymans
81558d6a94 gst/playback/: Add screenshot conversion code from totem.
Original commit message from CVS:
* gst/playback/Makefile.am:
* gst/playback/gstscreenshot.c: (feed_fakesrc), (save_result),
(create_element), (gst_play_frame_conv_convert):
* gst/playback/gstscreenshot.h:
Add screenshot conversion code from totem.
* gst/playback/gstplay-marshal.list:
* gst/playback/gstplaybin2.c: (gst_play_marshal_BUFFER__BOXED),
(gst_play_bin_class_init), (gst_play_bin_convert_frame),
(gst_play_bin_get_property), (no_more_pads_cb), (activate_group):
Implement frame property to get a color-unconverted snapshot.
Implement convert-frame action signal to get a converted snapshot image.
Configure connection speed in uridecodebin.
Document some more properties.
* gst/playback/gstplaysink.c: (gst_play_sink_class_init),
(gen_video_chain), (gen_audio_chain), (gst_play_sink_reconfigure),
(gst_play_sink_get_last_frame):
* gst/playback/gstplaysink.h:
Use last-buffer property of the video sink to get a video snapshot.
* tests/examples/seek/seek.c: (shot_cb), (main):
Add snapshot button for playbin2 and use the frame property to save the
frame as a png in the current directory.
2008-02-19 15:02:33 +00:00
Josep Torra Valles
58a9fd3622 gst/typefind/gsttypefindfunctions.c: Add typefinding support for h264 elementary streams.
Original commit message from CVS:
Patch by: Josep Torra Valles <josep at fluendo dot com>
* gst/typefind/gsttypefindfunctions.c: (h264_video_type_find),
(plugin_init):
Add typefinding support for h264 elementary streams.
Fixes bug #517420.
2008-02-19 11:45:56 +00:00
Stefan Kost
054842ca82 configure.ac: Require CVS of core for new API in collectpads.
Original commit message from CVS:
* configure.ac:
Require CVS of core for new API in collectpads.
* gst/adder/gstadder.c:
Use new API to make adder sparse stream aware.
2008-02-18 13:51:34 +00:00
Wim Taymans
5fc67f8bd3 gst/playback/gstplaybin2.c: Get the object data correct so that we can remove our channels correctly.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (pad_added_cb), (pad_removed_cb),
(no_more_pads_cb):
Get the object data correct so that we can remove our channels
correctly.
* gst/playback/gstplaysink.c: (gen_video_chain), (gen_audio_chain),
(gen_vis_chain), (gst_play_sink_reconfigure),
(gst_play_sink_request_pad):
Add option to disable async behaviour in the sinks when possible. This
makes it possible to avoid an audio queue when dealing with
visualisations.
Add option to add a queue for the audio path.
* tests/examples/seek/seek.c: (clear_streams), (update_streams),
(main):
Disable the vis checkbox to match the defaults of playbin2.
Only get the stream info when we need to.
2008-02-18 11:54:15 +00:00
Wim Taymans
5659831526 gst/playback/gstplaysink.c: Move tee in front of the audio and vis pipelines.
Original commit message from CVS:
* gst/playback/gstplaysink.c: (gst_play_sink_set_mute),
(gst_play_sink_get_mute), (gen_video_chain), (gen_audio_chain),
(gen_vis_chain), (gst_play_sink_reconfigure),
(gst_play_sink_request_pad):
Move tee in front of the audio and vis pipelines.
Add queue for audio for now.
Add visualisation support.
* tests/examples/seek/seek.c: (main):
Visualisation is by default disabled.
2008-02-15 18:38:52 +00:00
Wim Taymans
609daaede3 gst/playback/: Add mute property.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
(gst_play_bin_set_property), (gst_play_bin_get_property),
(pad_added_cb), (pad_removed_cb), (no_more_pads_cb):
* gst/playback/gstplaysink.c: (gst_play_sink_set_mute),
(gst_play_sink_get_mute), (gen_audio_chain):
* gst/playback/gstplaysink.h:
Add mute property.
* gst/playback/gststreamselector.c: (gst_selector_pad_event),
(gst_selector_pad_chain):
* gst/playback/gststreamselector.h:
Make sure we forward the event only once.
* tests/examples/seek/seek.c: (stop_cb), (mute_toggle_cb), (main):
Add and implement the mute button for playbin2.
2008-02-14 18:24:42 +00:00
Tim-Philipp Müller
1d9e1d6a3d gst/playback/: Handle case where we can't create the volume element a bit better (#514307).
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gen_audio_element):
* gst/playback/gstplaysink.c: (gen_audio_chain):
Handle case where we can't create the volume element a bit
better (#514307).
2008-02-11 18:31:43 +00:00
Tim-Philipp Müller
cfe66ed251 gst/typefind/gsttypefindfunctions.c: Bump rank of jpeg and png typefinders, which will return maximum probability in ...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c:
Bump rank of jpeg and png typefinders, which will return maximum
probability in the most common cases (thus short-circuiting more
expensive typefinders like the mp3 one for these two quite common
image types).
2008-02-11 13:03:13 +00:00
Zaheer Abbas Merali
b006ba7afe gst/playback/gstplaybasebin.c: Set is_dynamic as True if there are elements with both request and sometimes src pad t...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c:
Set is_dynamic as True if there are elements with both request
and sometimes src pad templates instead of breaking out when it
finds the first pad template that is a src.
2008-02-09 10:41:36 +00:00
Wim Taymans
c8bb67d0ca gst/playback/gstplay-marshal.list: Added marshal for streamselector Tags.
Original commit message from CVS:
* gst/playback/gstplay-marshal.list:
Added marshal for streamselector Tags.
* gst/playback/gstplaybasebin.c: (set_active_source):
Streamselector now selects pads based on the pad object instead of its
name.
* gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
(init_group), (gst_play_bin_init), (get_group), (get_tags),
(gst_play_bin_get_video_tags), (gst_play_bin_get_audio_tags),
(gst_play_bin_get_text_tags),
(gst_play_bin_set_current_video_stream),
(gst_play_bin_set_current_audio_stream),
(gst_play_bin_set_current_text_stream),
(gst_play_bin_set_property), (gst_play_bin_get_property),
(pad_added_cb), (pad_removed_cb), (autoplug_select_cb):
Remove option to mute streams with the current-a/v/t property, we have
this functionality in the flags.
Add signals to notify when the number of A/V/T channels changed.
Add action signals to get tags for the A/V/T streams.
Implement setting the current A/V/T stream.
Rearrange some things to simplify stream selection.
Implement volume.
* gst/playback/gstplaysink.c: (gst_play_sink_set_volume),
(gst_play_sink_get_volume), (gst_play_sink_set_property),
(gst_play_sink_get_property), (gen_video_chain), (gen_audio_chain),
(activate_vis), (gst_play_sink_reconfigure):
* gst/playback/gstplaysink.h:
Add and implement volume setting methods.
* gst/playback/gststreamselector.c: (gst_selector_pad_class_init),
(gst_selector_pad_finalize), (gst_selector_pad_get_property),
(gst_selector_pad_event), (gst_stream_selector_class_init),
(gst_stream_selector_init), (gst_stream_selector_finalize),
(gst_stream_selector_set_property),
(gst_stream_selector_get_property),
(gst_stream_selector_get_linked_pad),
(gst_stream_selector_request_new_pad):
* gst/playback/gststreamselector.h:
Add pad properties for tags and status of pads.
Keep tags on pads.
Make active pad selection based on pad object instead of name.
2008-02-08 17:47:37 +00:00
Wim Taymans
b5aaf1e1a9 gst/tcp/gstfdset.h: Remove unused field to same some memory.
Original commit message from CVS:
* gst/tcp/gstfdset.h:
Remove unused field to same some memory.
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
Mark action signals as such.
2008-02-06 15:07:30 +00:00
Wim Taymans
899330d904 gst/playback/gstplaybin2.c: Remove stream-info, we going for something easier.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
(get_group), (get_n_pads), (gst_play_bin_get_property),
(pad_added_cb), (no_more_pads_cb), (perform_eos),
(autoplug_select_cb), (deactivate_group):
Remove stream-info, we going for something easier.
Refactor getting the current group.
Implement getting the number of audio/video/text streams.
* gst/playback/gststreamselector.c:
(gst_stream_selector_class_init), (gst_stream_selector_init),
(gst_stream_selector_get_property),
(gst_stream_selector_request_new_pad),
(gst_stream_selector_release_pad):
* gst/playback/gststreamselector.h:
Add property for number of pads.
* tests/examples/seek/seek.c: (set_scale), (update_flag),
(vis_toggle_cb), (audio_toggle_cb), (video_toggle_cb),
(text_toggle_cb), (update_streams), (msg_async_done),
(msg_state_changed), (main):
Block slider callback when updating the slider position.
Add gui elements for controlling playbin2.
Add callback for async_done that updates position/duration.
2008-02-01 16:44:21 +00:00
David Schleef
5aad3658f8 gst/videoscale/vs_4tap.c: Fix valgrind error on 4tap scaling method.
Original commit message from CVS:
* gst/videoscale/vs_4tap.c:
Fix valgrind error on 4tap scaling method.
2008-01-14 01:19:34 +00:00
Tim-Philipp Müller
047fb95bad gst/playback/gstdecodebin.c: Make sure we error out correctly if we can't activate one of the elements we've added. ...
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (try_to_link_1):
Make sure we error out correctly if we can't activate one of
the elements we've added.  Fixes #508138.
2008-01-08 20:48:00 +00:00
Wim Taymans
9c9f60777a gst/playback/gstplay-enum.*: Add enums for configuration flags.
Original commit message from CVS:
* gst/playback/gstplay-enum.c:
(register_gst_autoplug_select_result),
(gst_autoplug_select_result_get_type), (register_gst_play_flags),
(gst_play_flags_get_type):
* gst/playback/gstplay-enum.h:
Add enums for configuration flags.
* gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
(init_group), (gst_play_bin_init), (gst_play_bin_set_property),
(gst_play_bin_get_property), (no_more_pads_cb),
(autoplug_select_cb), (gst_play_bin_change_state):
Merge mode with flags.
Add more property getters/setters, defaults and docs.
Add properties to get number of audio/video/text streams.
Create sink object in _init so that we can always rely on it being
there.
* gst/playback/gstplaysink.c: (gst_play_sink_init),
(gen_video_chain), (gen_audio_chain), (gen_vis_chain),
(activate_vis), (gst_play_sink_reconfigure),
(gst_play_sink_set_flags), (gst_play_sink_get_flags),
(gst_play_sink_change_state):
* gst/playback/gstplaysink.h:
Use flags to configure the sink pipelines.
Add tee before audio pipeline so that we can use it for visualisations.
Start working on integrating visualisations.
Remove mode, we can do everything with the flags now.
Add method to configue the sink pipeline.
2008-01-07 11:40:04 +00:00
Sebastian Dröge
3ac84ec4ff gst/volume/: Use GstAudioFilter as base class for the volume element instead of plain GstBaseTransform.
Original commit message from CVS:
* gst/volume/Makefile.am:
* gst/volume/gstvolume.c: (volume_choose_func),
(gst_volume_base_init), (gst_volume_class_init), (gst_volume_init),
(volume_setup):
* gst/volume/gstvolume.h:
Use GstAudioFilter as base class for the volume element instead of
plain GstBaseTransform.
2008-01-03 20:33:58 +00:00
Thijs Vermeir
b3739a8e7d gst/subparse/gstssaparse.c: combine if's
Original commit message from CVS:
* gst/subparse/gstssaparse.c:
combine if's
2007-12-29 20:55:39 +00:00
Thijs Vermeir
41cc98e287 gst/subparse/gstssaparse.c: remove duplicate log message
Original commit message from CVS:
* gst/subparse/gstssaparse.c:
remove duplicate log message
2007-12-29 19:23:59 +00:00
Wim Taymans
7cb7bffb9e gst/playback/gstplaybin2.c: Code cleanups.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
(gst_play_bin_finalize), (gst_play_bin_set_uri),
(gst_play_bin_set_suburi), (gst_play_bin_set_property),
(gst_play_bin_get_property), (pad_removed_cb), (drained_cb),
(autoplug_select_cb), (activate_group), (deactivate_group),
(setup_next_source), (save_current_group),
(gst_play_bin_change_state):
Code cleanups.
Remove next-uri, we can use the uri property just fine.
Fix some crasher.
Unref uridecodebin when switching.
Fix going to READY.
* gst/playback/gstplaysink.c: (gst_play_sink_class_init),
(gst_play_sink_init), (gst_play_sink_dispose),
(gst_play_sink_finalize), (gst_play_sink_vis_unblocked),
(gst_play_sink_vis_blocked), (gst_play_sink_set_video_sink),
(gst_play_sink_set_audio_sink), (gst_play_sink_set_vis_plugin),
(gst_play_sink_set_property), (gst_play_sink_get_property),
(gen_video_chain), (gen_text_element), (gen_audio_chain),
(gen_vis_element), (gst_play_sink_get_mode),
(gst_play_sink_set_mode), (gst_play_sink_set_flags),
(gst_play_sink_get_flags), (gst_play_sink_request_pad),
(gst_play_sink_release_pad), (gst_play_sink_send_event_to_sink),
(gst_play_sink_change_state):
* gst/playback/gstplaysink.h:
Add some locking to make things threadsafe.
* gst/playback/test7.c: (about_to_finish_cb):
Fix test.
2007-12-28 09:00:27 +00:00
Tim-Philipp Müller
bd01fd3a57 gst/videoscale/gstvideoscale.c: Don't claim to be able to handle/transform caps that can't really be handled by the c...
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c: (gst_video_scale_set_property),
(gst_video_scale_get_property), (gst_video_scale_transform_caps),
(gst_video_scale_transform):
Don't claim to be able to handle/transform caps that can't really
be handled by the currently selected scaling method (here: RGB or
packed YUV with 4-tap method). Also add locking to method property.
* tests/check/pipelines/simple-launch-lines.c: (setup_pipeline),
(test_basetransform_based):
Some test pipelines for the above (not entirely valgrind clean yet
apparently).
2007-12-22 12:06:47 +00:00
Tim-Philipp Müller
032e064516 gst/playback/gststreamselector.c: Don't leak event.
Original commit message from CVS:
* gst/playback/gststreamselector.c: (gst_selector_pad_event):
Don't leak event.
2007-12-21 22:26:47 +00:00
Tim-Philipp Müller
377bde7868 gst/playback/.cvsignore: Ignore more.
Original commit message from CVS:
* gst/playback/.cvsignore:
Ignore more.
2007-12-20 17:13:37 +00:00
Tim-Philipp Müller
85f189aee5 Make switching off of subtitles work. To avoid all kind of problems with unlinking of the subtitle input, we just kee...
Original commit message from CVS:
* ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
* gst/playback/gstplaybasebin.c: (set_subtitles_visible),
(set_active_source):
* gst/playback/gstplaybasebin.h:
* gst/playback/gstplaybin.c: (gst_play_bin_class_init),
(setup_sinks), (playbin_set_subtitles_visible):
Make switching off of subtitles work. To avoid all kind of
problems with unlinking of the subtitle input, we just keep
the subtitle inputs linked as they are and tell textoverlay
not to render them. Fixes #373011.
Other subtitle switching issues (esp. when there are both
external and in-stream subtitles) remain. They'll be solved
in playbin2.
2007-12-20 10:41:29 +00:00
Wim Taymans
e56165db4f gst/playback/gststreamselector.c: Init the pad segment too.
Original commit message from CVS:
* gst/playback/gststreamselector.c: (gst_selector_pad_init):
Init the pad segment too.
2007-12-18 16:21:35 +00:00
David Schleef
c5b66243be gst/videotestsrc/gstvideotestsrc.*: Add a "blink" pattern. Turn on the pain. Apologies. It's useful for testing ve...
Original commit message from CVS:
* gst/videotestsrc/gstvideotestsrc.c:
* gst/videotestsrc/gstvideotestsrc.h:
Add a "blink" pattern.  Turn on the pain.  Apologies.  It's useful
for testing vertical refresh synchronization.
2007-12-18 01:01:23 +00:00
Sebastian Dröge
248742277c Use new gst_base_transform_set_gap_aware() function as volume correctly handles GST_BUFFER_FLAG_GAP. Require core 0.1...
Original commit message from CVS:
* configure.ac:
* gst/volume/gstvolume.c: (gst_volume_init):
Use new gst_base_transform_set_gap_aware() function as volume
correctly handles GST_BUFFER_FLAG_GAP. Require core 0.10.15.1
for this.
2007-12-15 03:40:34 +00:00
Wim Taymans
671d766d8a gst/playback/gstqueue2.c: Use separate timers for input and output rates.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (gst_queue_init), (gst_queue_finalize),
(reset_rate_timer), (update_in_rates), (update_out_rates),
(gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
(gst_queue_chain), (gst_queue_loop):
Use separate timers for input and output rates.
Pause measuring the output rate when we block for more data.
See #503262.
2007-12-14 18:46:12 +00:00
Christian Schaller
9153699f65 update spec file and add two missing files for disting
Original commit message from CVS:
update spec file and add two missing files for disting
2007-12-14 16:23:06 +00:00
Wim Taymans
2da1bb2538 gst/playback/gstqueue2.c: Pause the timer to measure the input rate when we block because the queue is filled. See #5...
Original commit message from CVS:
* gst/playback/gstqueue2.c: (gst_queue_chain):
Pause the timer to measure the input rate when we block because the
queue is filled. See #503262.
2007-12-14 09:24:55 +00:00
Wim Taymans
74e5172181 gst/playback/gstdecodebin2.c: Expose the right pad in the right place with the right element.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (analyze_new_pad), (connect_pad):
Expose the right pad in the right place with the right element.
2007-12-13 12:31:38 +00:00
Wim Taymans
802b38c200 gst/audioconvert/Makefile.am: Also link to libm.
Original commit message from CVS:
* gst/audioconvert/Makefile.am:
Also link to libm.
2007-12-10 15:21:41 +00:00
Robin Stocker
7bbbf15ad8 gst/subparse/gstsubparse.c: Some .srt files start with chunk number 0 and not chunk number 1, recognise and accept th...
Original commit message from CVS:
Patch by: Robin Stocker <robin dot stocker at gmx dot ch>
* gst/subparse/gstsubparse.c: (gst_sub_parse_data_format_autodetect):
Some .srt files start with chunk number 0 and not chunk number 1,
recognise and accept those as well (fixes #502497).
* tests/check/elements/subparse.c: (srt_input), (srt_input0),
(test_src):
Add unit test for the above.
2007-12-08 18:38:39 +00:00
Wim Taymans
356971158c gst/playback/gstplay-enum.*: Add missing files.
Original commit message from CVS:
* gst/playback/gstplay-enum.c:
(register_gst_autoplug_select_result),
(gst_autoplug_select_result_get_type):
* gst/playback/gstplay-enum.h:
Add missing files.
2007-12-06 12:08:21 +00:00
Wim Taymans
f2f9bf045b gst/playback/Makefile.am: Group decodebin2 and uridecodebin into the same plugin so that they can share the GEnumType.
Original commit message from CVS:
* gst/playback/Makefile.am:
Group decodebin2 and uridecodebin into the same plugin so that they
can share the GEnumType.
* gst/playback/gstdecodebin2.c: (_gst_array_accumulator),
(_gst_select_accumulator), (gst_decode_bin_class_init),
(gst_decode_bin_init), (gst_decode_bin_autoplug_sort),
(gst_decode_bin_autoplug_select), (gst_decode_bin_autoplug_add),
(analyze_new_pad), (connect_pad), (gst_decode_bin_plugin_init):
Add signal to sort factories instead of the more awkward autoplug-select
signal.
Modify autoplug_select so that we can try, skip or expose the
autopluggin of an element on a pad.
* gst/playback/gstfactorylists.c: (compare_ranks),
(decoders_filter), (sinks_filter), (gst_factory_list_is_type),
(element_filter), (gst_factory_list_get_elements),
(gst_factory_list_debug), (gst_factory_list_filter):
* gst/playback/gstfactorylists.h:
Simplify the API, allow getting elements based on mask.
* gst/playback/gstplay-marshal.list:
Add some more marshallers.
* gst/playback/gstplaybin2.c: (init_group), (gst_play_bin_init),
(gst_play_bin_finalize), (pad_removed_cb), (autoplug_factories_cb),
(autoplug_select_cb), (activate_group):
Add support for managing non-raw sinks by providing a custom element and
sink list to decodebin2.
Try to plug non-raw sinks when decodebin2 using autoplug-select of
decodebin2.
* gst/playback/gstplaysink.c: (gen_video_chain), (gen_audio_chain),
(gst_play_sink_set_mode), (gst_play_sink_request_pad):
* gst/playback/gstplaysink.h:
Add support for raw and non-raw sinks.
Add support to force sinks selected by playbin2.
Don't plug raw converters for non-raw sinks.
* gst/playback/gsturidecodebin.c: (_gst_array_accumulator),
(_gst_select_accumulator), (gst_uri_decode_bin_class_init),
(proxy_autoplug_select_signal), (gst_uri_decode_bin_plugin_init),
(plugin_init):
Use right accumulators.
Proxy new signal.
2007-12-05 17:11:48 +00:00
Wim Taymans
11bf488b85 gst/playback/: Refactor some common code to filter factories and check caps compat.
Original commit message from CVS:
* gst/playback/Makefile.am:
* gst/playback/gstfactorylists.c: (compare_ranks), (print_feature),
(get_feature_array), (decoders_filter), (sinks_filter),
(gst_factory_list_get_decoders), (gst_factory_list_get_sinks),
(gst_factory_list_filter):
* gst/playback/gstfactorylists.h:
Refactor some common code to filter factories and check caps compat.
* gst/playback/gstdecodebin.c:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
(gst_decode_bin_init), (gst_decode_bin_dispose),
(gst_decode_bin_autoplug_continue),
(gst_decode_bin_autoplug_factories),
(gst_decode_bin_autoplug_select), (analyze_new_pad),
(find_compatibles):
* gst/playback/gstplaybin.c:
* gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
(gst_play_bin_init), (gst_play_bin_finalize),
(autoplug_factories_cb), (activate_group):
* gst/playback/gstqueue2.c:
* gst/playback/gsturidecodebin.c: (proxy_unknown_type_signal),
(proxy_autoplug_continue_signal),
(proxy_autoplug_factories_signal), (proxy_autoplug_select_signal),
(proxy_drained_signal):
Add some more debug info and use factor filtering code.
2007-11-30 17:47:15 +00:00
Julien Moutte
0625160416 configure.ac: Add QuickTime Wrapper plug-in.
Original commit message from CVS:
2007-11-26  Julien Moutte  <julien@fluendo.com>

* configure.ac: Add QuickTime Wrapper plug-in.
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_push_drain), (gst_speex_resample_process): Fix
build on Mac OS X Leopard. Incorrect printf format arguments.
* sys/Makefile.am:
* sys/qtwrapper/Makefile.am:
* sys/qtwrapper/audiodecoders.c:
(qtwrapper_audio_decoder_base_init),
(qtwrapper_audio_decoder_class_init),
(qtwrapper_audio_decoder_init),
(clear_AudioStreamBasicDescription), (fill_indesc_mp3),
(fill_indesc_aac), (fill_indesc_samr), (fill_indesc_generic),
(make_samr_magic_cookie), (open_decoder),
(qtwrapper_audio_decoder_sink_setcaps), (process_buffer_cb),
(qtwrapper_audio_decoder_chain),
(qtwrapper_audio_decoder_sink_event),
(qtwrapper_audio_decoders_register):
* sys/qtwrapper/codecmapping.c: (audio_caps_from_string),
(fourcc_to_caps):
* sys/qtwrapper/codecmapping.h:
* sys/qtwrapper/imagedescription.c: (image_description_for_avc1),
(image_description_for_mp4v), (image_description_from_stsd_buffer),
(image_description_from_codec_data):
* sys/qtwrapper/imagedescription.h:
* sys/qtwrapper/qtutils.c: (get_name_info_from_component),
(get_output_info_from_component), (dump_avcc_atom),
(dump_image_description), (dump_codec_decompress_params),
(addSInt32ToDictionary), (dump_cvpixel_buffer),
(DestroyAudioBufferList), (AllocateAudioBufferList):
* sys/qtwrapper/qtutils.h:
* sys/qtwrapper/qtwrapper.c: (plugin_init):
* sys/qtwrapper/qtwrapper.h:
* sys/qtwrapper/videodecoders.c:
(qtwrapper_video_decoder_base_init),
(qtwrapper_video_decoder_class_init),
(qtwrapper_video_decoder_init), (qtwrapper_video_decoder_finalize),
(fill_image_description), (new_image_description), (close_decoder),
(open_decoder), (qtwrapper_video_decoder_sink_setcaps),
(decompressCb), (qtwrapper_video_decoder_chain),
(qtwrapper_video_decoder_sink_event),
(qtwrapper_video_decoders_register): Initial import of QuickTime
wrapper jointly developped by Songbird authors (Pioneers of the
Inevitable) and Fluendo.
2007-11-26 13:19:46 +00:00
Stefan Kost
1cfef609d0 gst/: Add GAP-flag support.
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
* gst/volume/gstvolume.c:
* gst/volume/gstvolume.h:
Add GAP-flag support.
2007-11-26 12:25:55 +00:00
Sebastian Dröge
dacc06a547 gst/speexresample/: Update speex resampler to latest SVN. We're now down to only the changes noted in README again.
Original commit message from CVS:
* gst/speexresample/README:
* gst/speexresample/arch.h:
* gst/speexresample/resample.c: (resampler_basic_direct_single),
(resampler_basic_direct_double),
(resampler_basic_interpolate_single),
(resampler_basic_interpolate_double),
(speex_resampler_process_native), (speex_resampler_process_float),
(speex_resampler_process_int),
(speex_resampler_process_interleaved_float),
(speex_resampler_process_interleaved_int),
(speex_resampler_get_input_latency),
(speex_resampler_get_output_latency):
* gst/speexresample/speex_resampler.h:
Update speex resampler to latest SVN. We're now down to only the
changes noted in README again.
* gst/speexresample/speex_resampler_wrapper.h:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_push_drain), (gst_speex_resample_query):
Adjust to API changes.
2007-11-26 08:43:25 +00:00
Sebastian Dröge
155d1b123d gst/speexresample/gstspeexresample.c: Only post the latency message if we have a resampler state already.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_update_state):
Only post the latency message if we have a resampler state already.
2007-11-23 10:21:31 +00:00
Sebastian Dröge
8edd45dbde gst/audioresample/gstaudioresample.c: Implement latency query.
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
(audioresample_query), (audioresample_query_type),
(gst_audioresample_set_property):
Implement latency query.
2007-11-23 10:21:11 +00:00
Sebastian Dröge
816466b67f gst/speexresample/gstspeexresample.c: Also post GST_MESSAGE_LATENCY if the latency changes.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_update_state):
Also post GST_MESSAGE_LATENCY if the latency changes.
2007-11-23 10:01:33 +00:00
Sebastian Dröge
f564ebf8cb gst/speexresample/: Add functions to push the remaining samples and to get the latency of the resampler. These will g...
Original commit message from CVS:
* gst/speexresample/resample.c: (speex_resampler_get_latency),
(speex_resampler_drain_float), (speex_resampler_drain_int),
(speex_resampler_drain_interleaved_float),
(speex_resampler_drain_interleaved_int):
* gst/speexresample/speex_resampler.h:
* gst/speexresample/speex_resampler_wrapper.h:
Add functions to push the remaining samples and to get the latency
of the resampler. These will get added to Speex SVN in this or a
slightly changed form at some point too and should get merged then
again.
* gst/speexresample/gstspeexresample.c: (gst_speex_resample_init),
(gst_speex_resample_init_state),
(gst_speex_resample_transform_size),
(gst_speex_resample_push_drain), (gst_speex_resample_event),
(gst_speex_fix_output_buffer), (gst_speex_resample_process),
(gst_speex_resample_query), (gst_speex_resample_query_type):
Drop the prepending zeroes and output the remaining samples on EOS.
Also properly implement the latency query for this. speexresample
should be completely ready for production use now.
2007-11-23 08:48:50 +00:00
Sebastian Dröge
d834d1cb48 gst/speexresample/README: Add README explaining where the resampling code was taken from and which changes were done.
Original commit message from CVS:
* gst/speexresample/README:
Add README explaining where the resampling code was taken from
and which changes were done.
* gst/speexresample/resample.c: (speex_alloc), (speex_realloc),
(speex_free):
Use g_malloc() and friends instead of malloc() to achieve higher
portability and define the functions inline.
* gst/speexresample/speex_resampler.h:
Add back some useless preprocessor stuff to keep the diff between
our version and the one from the Speex SVN repository lower.
2007-11-21 10:18:56 +00:00
Sebastian Dröge
d832d9bb16 gst/speexresample/gstspeexresample.c: Some small cleanup and addition of a TODO item.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_fix_output_buffer), (gst_speex_resample_transform):
Some small cleanup and addition of a TODO item.
2007-11-20 20:23:25 +00:00
Sebastian Dröge
25c4adab31 gst/speexresample/Makefile.am: Add missing file.
Original commit message from CVS:
* gst/speexresample/Makefile.am:
Add missing file.
2007-11-20 12:56:00 +00:00
Sebastian Dröge
66f2838c8c Add speexresample to the docs and while at that do a make update.
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.signals:
* docs/plugins/inspect/plugin-bz2.xml:
* docs/plugins/inspect/plugin-cdxaparse.xml:
* docs/plugins/inspect/plugin-dtsdec.xml:
* docs/plugins/inspect/plugin-equalizer.xml:
* docs/plugins/inspect/plugin-faac.xml:
* docs/plugins/inspect/plugin-faad.xml:
* docs/plugins/inspect/plugin-filter.xml:
* docs/plugins/inspect/plugin-freeze.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gsm.xml:
* docs/plugins/inspect/plugin-gstrtpmanager.xml:
* docs/plugins/inspect/plugin-h264parse.xml:
* docs/plugins/inspect/plugin-modplug.xml:
* docs/plugins/inspect/plugin-mpeg2enc.xml:
* docs/plugins/inspect/plugin-musepack.xml:
* docs/plugins/inspect/plugin-musicbrainz.xml:
* docs/plugins/inspect/plugin-nsfdec.xml:
* docs/plugins/inspect/plugin-replaygain.xml:
* docs/plugins/inspect/plugin-soundtouch.xml:
* docs/plugins/inspect/plugin-spcdec.xml:
* docs/plugins/inspect/plugin-spectrum.xml:
* docs/plugins/inspect/plugin-speed.xml:
* docs/plugins/inspect/plugin-tta.xml:
* docs/plugins/inspect/plugin-videosignal.xml:
* docs/plugins/inspect/plugin-xingheader.xml:
* docs/plugins/inspect/plugin-xvid.xml:
* gst/speexresample/gstspeexresample.h:
Add speexresample to the docs and while at that do a make update.
2007-11-20 07:47:27 +00:00
Sebastian Dröge
3adf5a8875 gst/speexresample/gstspeexresample.c: If the resampler gives less output samples than expected adjust the output buff...
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_fix_output_buffer), (gst_speex_resample_process):
If the resampler gives less output samples than expected
adjust the output buffer and print a warning.
2007-11-20 07:30:30 +00:00
Sebastian Dröge
7fc30c9d28 Add resample element based on the Speex resampling algorithm.
Original commit message from CVS:
* configure.ac:
* gst/speexresample/arch.h:
* gst/speexresample/fixed_generic.h:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_base_init), (gst_speex_resample_class_init),
(gst_speex_resample_init), (gst_speex_resample_start),
(gst_speex_resample_stop), (gst_speex_resample_get_unit_size),
(gst_speex_resample_transform_caps),
(gst_speex_resample_init_state), (gst_speex_resample_update_state),
(gst_speex_resample_reset_state), (gst_speex_resample_parse_caps),
(gst_speex_resample_transform_size), (gst_speex_resample_set_caps),
(gst_speex_resample_event), (gst_speex_resample_check_discont),
(gst_speex_resample_process), (gst_speex_resample_transform),
(gst_speex_resample_set_property),
(gst_speex_resample_get_property), (plugin_init):
* gst/speexresample/gstspeexresample.h:
* gst/speexresample/resample.c: (speex_alloc), (speex_realloc),
(speex_free), (compute_func), (main), (sinc), (cubic_coef),
(resampler_basic_direct_single), (resampler_basic_direct_double),
(resampler_basic_interpolate_single),
(resampler_basic_interpolate_double), (update_filter),
(speex_resampler_init), (speex_resampler_init_frac),
(speex_resampler_destroy), (speex_resampler_process_native),
(speex_resampler_process_float), (speex_resampler_process_int),
(speex_resampler_process_interleaved_float),
(speex_resampler_process_interleaved_int),
(speex_resampler_set_rate), (speex_resampler_get_rate),
(speex_resampler_set_rate_frac), (speex_resampler_get_ratio),
(speex_resampler_set_quality), (speex_resampler_get_quality),
(speex_resampler_set_input_stride),
(speex_resampler_get_input_stride),
(speex_resampler_set_output_stride),
(speex_resampler_get_output_stride), (speex_resampler_skip_zeros),
(speex_resampler_reset_mem), (speex_resampler_strerror):
* gst/speexresample/speex_resampler.h:
* gst/speexresample/speex_resampler_float.c:
* gst/speexresample/speex_resampler_int.c:
* gst/speexresample/speex_resampler_wrapper.h:
Add resample element based on the Speex resampling algorithm.
2007-11-20 07:02:45 +00:00
Stefan Kost
84b35b401a gst/playback/: Fix the build + little README update.
Original commit message from CVS:
* gst/playback/README:
* gst/playback/test7.c:
Fix the build + little README update.
2007-11-17 15:25:15 +00:00
Wim Taymans
b75b5525da gst/playback/: Add playbin2.
Original commit message from CVS:
* gst/playback/Makefile.am:
* gst/playback/gstplayback.c: (plugin_init):
* gst/playback/test7.c: (update_scale), (warning_cb), (error_cb),
(eos_cb), (about_to_finish_cb), (main):
Add playbin2.
Added gapless playback example.
* gst/playback/gstplaybasebin.c:
* gst/playback/gstplaybasebin.h:
* gst/playback/gstplaybin.c: (gst_play_bin_plugin_init):
* gst/playback/gstqueue2.c:
* gst/playback/test.c:
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
(pad_removed_cb):
* gst/playback/gststreaminfo.h:
Change email.
* gst/playback/gstplaybin2.c: (gst_play_bin_get_type),
(gst_play_bin_class_init), (init_group), (gst_play_bin_init),
(gst_play_bin_dispose), (gst_play_bin_set_uri),
(gst_play_bin_set_suburi), (gst_play_bin_set_property),
(gst_play_bin_get_property), (gst_play_bin_handle_message),
(pad_added_cb), (pad_removed_cb), (no_more_pads_cb), (perform_eos),
(drained_cb), (unlink_group), (activate_group),
(setup_next_source), (gst_play_bin_change_state),
(gst_play_bin2_plugin_init):
Added raw first version of playbin2. Does chained oggs and gapless
playback fine. No support for raw sinks yet. No visualisations or
subtitles yet.
* gst/playback/gstplaysink.c: (gst_play_sink_get_type),
(gst_play_sink_class_init), (gst_play_sink_init),
(gst_play_sink_dispose), (gst_play_sink_vis_unblocked),
(gst_play_sink_vis_blocked), (gst_play_sink_set_video_sink),
(gst_play_sink_set_audio_sink), (gst_play_sink_set_vis_plugin),
(gst_play_sink_set_property), (gst_play_sink_get_property),
(post_missing_element_message), (free_chain), (add_chain),
(activate_chain), (gen_video_chain), (gen_text_element),
(gen_audio_chain), (gen_vis_element), (gst_play_sink_get_mode),
(gst_play_sink_set_mode), (gst_play_sink_request_pad),
(gst_play_sink_release_pad), (gst_play_sink_send_event_to_sink),
(gst_play_sink_send_event), (gst_play_sink_change_state):
* gst/playback/gstplaysink.h:
Added Element that abstracts the sinks and their pipelines for playbin2.
2007-11-16 15:44:48 +00:00
Wim Taymans
3f43bfacd2 gst/playback/gststreamselector.*: Improve streamselector, make it select and unselect the current pad more intelligen...
Original commit message from CVS:
* gst/playback/gststreamselector.c: (gst_selector_pad_get_type),
(gst_selector_pad_class_init), (gst_selector_pad_init),
(gst_selector_pad_finalize), (gst_selector_pad_reset),
(gst_selector_pad_get_linked_pads), (gst_selector_pad_event),
(gst_selector_pad_getcaps), (gst_selector_pad_bufferalloc),
(gst_selector_pad_chain), (gst_stream_selector_get_type),
(gst_stream_selector_base_init), (gst_stream_selector_class_init),
(gst_stream_selector_init), (gst_stream_selector_set_property),
(gst_stream_selector_get_linked_pad),
(gst_stream_selector_getcaps),
(gst_stream_selector_is_active_sinkpad),
(gst_stream_selector_activate_sinkpad),
(gst_stream_selector_get_linked_pads),
(gst_stream_selector_request_new_pad),
(gst_stream_selector_release_pad):
* gst/playback/gststreamselector.h:
Improve streamselector, make it select and unselect the current pad more
intelligently.
Subclass GstPad for the sinkpads of the selector.
Handle segments more correctly.
Fix caps negotiation.
Implement release_pad.
2007-11-16 15:05:07 +00:00
Wim Taymans
0df5f5b2e6 gst/playback/gstdecodebin2.c: Add drained signal fired when decodebin finishes decoding the data.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
(gst_decode_group_check_if_drained), (source_pad_event_probe),
(remove_fakesink):
Add drained signal fired when decodebin finishes decoding the data.
Remove deprecated STATE_DIRTY message.
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
(unknown_type_cb), (new_decoded_pad_cb), (pad_removed_cb),
(analyse_source), (proxy_drained_signal), (make_decoder),
(source_new_pad), (value_list_append_structure_list),
(handle_redirect_message), (handle_message):
Proxy the new drained signal.
Handle pad removed from decodebin.
Handle redirect messages by sorting multiple redirections based on the
connection speed.
2007-11-16 12:51:44 +00:00
Stefan Kost
1c2fae7f8b gst/playback/gstdecodebin2.c: Dont leak ghostpad. Fixes #475451.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c:
Dont leak ghostpad. Fixes #475451.
2007-11-09 15:54:45 +00:00
Wim Taymans
905945738d Update some more docs and comments.
Original commit message from CVS:
* docs/design/design-decodebin.txt:
* gst/playback/gstdecodebin2.c: (analyze_new_pad):
Update some more docs and comments.
2007-11-09 12:21:52 +00:00
Tim-Philipp Müller
750a724841 gst/playback/gstplaybasebin.c: Avoid crash when there are external subtitles (fixes #491722).
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (queue_threshold_reached),
(finish_source):
Avoid crash when there are external subtitles (fixes #491722).
2007-11-06 11:09:30 +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
Tim-Philipp Müller
5861f366a0 gst/audioconvert/gstaudioconvert.c: Preserve channel layout when fixating the number of channels in the output caps, ...
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (find_suitable_channel_layout),
(gst_audio_convert_fixate_channels), (gst_audio_convert_fixate_caps):
Preserve channel layout when fixating the number of channels in the
output caps, or make sure there's a suitable channel position layout
set on the caps if required. Fixes #430677.
2007-10-31 17:54:48 +00:00
Tim-Philipp Müller
4c0e44de0f gst/playback/: Post nice/more useful error message if we don't have a decoder for the primary type.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (close_pad_link), (type_found):
* gst/playback/gstdecodebin2.c: (analyze_new_pad):
Post nice/more useful error message if we don't have a decoder for
the primary type.
2007-10-30 15:54:46 +00:00
Wim Taymans
b55c61c933 gst/playback/gstdecodebin2.c: Be a bit more useful, unblock the pads after we fired the no-more-pads signal so that w...
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_group_expose):
Be a bit more useful, unblock the pads after we fired the no-more-pads
signal so that we can use the signal to inspect and connect all pads
without having to keep extra state outside of decodebin.
2007-10-30 15:07:58 +00:00
Wim Taymans
b68d48e6bd gst/playback/gsturidecodebin.c: Implement default signal handler so that we return TRUE when nothing is connected.
Original commit message from CVS:
* gst/playback/gsturidecodebin.c:
(gst_uri_decode_bin_autoplug_continue),
(gst_uri_decode_bin_class_init), (no_more_pads_full):
Implement default signal handler so that we return TRUE when nothing is
connected.
2007-10-30 15:00:06 +00:00
Wim Taymans
8c20347774 gst/playback/gstdecodebin2.c: Move subtitle encoding property to decodebin2 so that it can set the property value on ...
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
(gst_decode_bin_dispose), (gst_decode_bin_set_caps),
(gst_decode_bin_set_subs_encoding),
(gst_decode_bin_get_subs_encoding), (gst_decode_bin_set_property),
(gst_decode_bin_get_property), (analyze_new_pad):
Move subtitle encoding property to decodebin2 so that it can set the
property value on all elements that it autoplugs and that require it.
Make caps refcounting more consistent in get/set.
* gst/playback/gsturidecodebin.c: (_gst_boolean_accumulator),
(gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
(gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_property),
(gst_uri_decode_bin_get_property), (proxy_unknown_type_signal),
(proxy_autoplug_continue_signal),
(proxy_autoplug_factories_signal), (proxy_autoplug_select_signal),
(make_decoder):
Proxy properties and relevant signals from the internal decodebin.
Make properties MT safe.
2007-10-25 17:36:49 +00:00
Wim Taymans
77cef56895 gst/playback/: Remove the autoplug-sort signal and replace it with a binding friendly autoplug-select signal.
Original commit message from CVS:
Inspired by patch of: René Stadler <mail at renestadler dot de>
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
(gst_decode_bin_autoplug_continue),
(gst_decode_bin_autoplug_factories),
(gst_decode_bin_autoplug_select), (analyze_new_pad), (connect_pad),
(find_compatibles):
* gst/playback/gstplay-marshal.list:
Remove the autoplug-sort signal and replace it with a binding friendly
autoplug-select signal.
Add an autoplug-factories signal that can be used to generate a list of
factories to try to autoplug.
Add the GstPad to the autoplugging signal args as it might be needed to
make a good factory selection.
Fix up the marshallers for this. Fixes #407282.
2007-10-24 11:07:57 +00:00
Wim Taymans
d33d2be0ed gst/playback/gstdecodebin.c: Make the window for a race in typefind and shutting down smaller until we figure out the...
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (new_pad), (type_found):
Make the window for a race in typefind and shutting down smaller until
we figure out the right locking here. Avoids #485753 usually.
* gst/playback/gstdecodebin2.c: (type_found), (pad_added_group_cb):
Remove unneeded lock causing a race in typefind and shutting down.
Fixes #485753.
* gst/playback/gstplaybin.c: (gst_play_bin_change_state):
Also remove sinks when going to NULL because we might not complete the
state change to PAUSED, causing the PAUSED->READY state change not to
happen.
2007-10-16 16:48:38 +00:00
Wim Taymans
b6a80a4e42 gst/playback/gstqueue2.c: Fix queue negotiation. See #486758.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (gst_queue_init), (gst_queue_push_one):
Fix queue negotiation. See #486758.
2007-10-15 11:38:39 +00:00
Wim Taymans
d0897a3528 gst/playback/: Don't disconnect the have_type signal because we never reconnect it later on. Instead keep a variable ...
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (type_found),
(gst_decode_bin_change_state):
* gst/playback/gstdecodebin2.c: (type_found),
(gst_decode_bin_change_state):
Don't disconnect the have_type signal because we never reconnect it
later on. Instead keep a variable to see if we already detected a type.
2007-10-08 17:12:32 +00:00
Wim Taymans
ecb6c19729 gst/playback/: Unlink the signal handler when we found the type, we're not going to do anything sensible with more ty...
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (add_fakesink), (type_found):
* gst/playback/gstdecodebin2.c: (gst_decode_bin_init),
(type_found):
Unlink the signal handler when we found the type, we're not going to do
anything sensible with more type_found signals anyway.
2007-10-08 10:47:26 +00:00
Wim Taymans
818434b664 gst/typefind/gsttypefindfunctions.c: Add typefind function for application/sdp.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (utf8_type_find),
(sdp_check_header), (sdp_type_find), (plugin_init):
Add typefind function for application/sdp.
Remove some old dirac typefind code that was ifdeffed out.
2007-10-01 10:22:46 +00:00
Wim Taymans
9f04b80b90 gst/playback/gstqueue2.c: Fix compilation wrt printf arguments.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (gst_queue_push_one):
Fix compilation wrt printf arguments.
2007-09-21 14:37:26 +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
d133f1548e gst/playback/gstqueue2.c: Also fix #476514 for queue2.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (update_buffering),
(gst_queue_locked_flush), (gst_queue_locked_enqueue),
(gst_queue_handle_sink_event), (gst_queue_chain),
(gst_queue_push_one), (gst_queue_sink_activate_push),
(gst_queue_src_activate_push), (gst_queue_src_activate_pull):
Also fix #476514 for queue2.
2007-09-17 16:22:17 +00:00
Julien Moutte
87f2e70427 gst/typefind/gsttypefindfunctions.c: Add some typefind for QCP files (RFC #3625)
Original commit message from CVS:
2007-09-14  Julien MOUTTE  <julien@moutte.net>

* gst/typefind/gsttypefindfunctions.c: (plugin_init): Add some
typefind for QCP files (RFC #3625)
2007-09-14 10:42:00 +00:00
Josep Torra Valles
1004fb0603 gst/playback/gstplaybasebin.c: Increase upper limit for audio queue a bit; fixes preroll problem with playbin and dec...
Original commit message from CVS:
Patch by: Josep Torra Valles <josep@fluendo.com>
* gst/playback/gstplaybasebin.c:
Increase upper limit for audio queue a bit; fixes preroll problem
with playbin and decodebin2 when playing a quicktime trailer with
multichannel audio via http (#464666).
2007-09-11 11:29:12 +00:00
Stefan Kost
3df6b8ad42 gst/playback/gstdecodebin2.c: Don't leak request pads. Fixes #475395.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c:
Don't leak request pads. Fixes #475395.
2007-09-10 12:05:34 +00:00
Sebastian Dröge
6fa7788c5d Revert the latest change: floating point samples are allowed to have any value, not only values in the range [-1,1]. ...
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_choose_func):
* tests/check/elements/volume.c: (GST_START_TEST):
Revert the latest change: floating point samples are allowed to
have any value, not only values in the range [-1,1]. Thanks to Andy
Wingo for noticing.
Also fix processing of int32 samples with volumes > 4 by making the
unity value smaller which prevents overflows.
2007-09-09 04:08:48 +00:00
Sebastian Dröge
6d7debb0bb gst/volume/gstvolume.c: Correctly clamp float/double samples in the [-1.0,1.0] range to prevent weird effects.
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_choose_func),
(volume_process_double), (volume_process_double_clamp),
(volume_process_float_clamp):
Correctly clamp float/double samples in the [-1.0,1.0] range to
prevent weird effects.
* tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
Add unit tests for all samples types that had none before.
2007-09-05 21:20:12 +00:00
Tim-Philipp Müller
12728158b5 gst/playback/gststreaminfo.c: Fix build.
Original commit message from CVS:
* gst/playback/gststreaminfo.c:
Fix build.
2007-09-05 14:01:25 +00:00
Stefan Kost
53c6315b6b gst/playback/gststreaminfo.c: Clean up some half-disabled code and comment.
Original commit message from CVS:
* gst/playback/gststreaminfo.c:
Clean up some half-disabled code and comment.
2007-09-05 10:32:09 +00:00
Johan Dahlin
417107b40e gst/typefind/gsttypefindfunctions.c (plugin_init): Add an audio/x-nsf typefind function for the nsfdec element.
Original commit message from CVS:
2007-09-03  Johan Dahlin  <jdahlin@async.com.br>

* gst/typefind/gsttypefindfunctions.c (plugin_init):
Add an audio/x-nsf typefind function for the nsfdec element.
2007-09-04 01:50:55 +00:00
Renato Filho
ac042e8869 gst/playback/gstplaybasebin.c: Included "myth://" on stream_uris list for enable buffering to mythtv files
Original commit message from CVS:
* gst/playback/gstplaybasebin.c:
Included "myth://" on stream_uris list for enable buffering to mythtv files
2007-09-03 20:46:38 +00:00
Stefan Kost
d2d03ba2f6 The tcp and subparse plugins are under gst, but not totaly free of dependencies. Handle selection inconfigure.ac, so ...
Original commit message from CVS:
* configure.ac:
* gst/Makefile.am:
The tcp and subparse plugins are under gst, but not totaly free of
dependencies. Handle selection inconfigure.ac, so that they show up
on the final list of what is build and what is not. Maybe they should
better be moved to ext.
2007-08-30 07:29:55 +00:00
Daniel Díaz
b2f2cfc132 Check if libxml provides HTML parser which subparse needs.
Original commit message from CVS:
Patch by: Daniel Díaz  <yosoy@danieldiaz.org>
* configure.ac:
* gst/Makefile.am:
Check if libxml provides HTML parser which subparse needs.
Fixes #451970.
2007-08-30 06:58:46 +00:00
Tim-Philipp Müller
bed6719df7 gst/subparse/gstssaparse.c: Convert SSA newline codes into actual newline characters (#470766).
Original commit message from CVS:
* gst/subparse/gstssaparse.c:
Convert SSA newline codes into actual newline characters (#470766).
2007-08-29 12:16:46 +00:00
Jan Schmidt
973bbf88af gst/playback/gstdecodebin.c: We need to set up delayed-linking whenever the caps are non-fixed, not just when there a...
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (close_pad_link):
We need to set up delayed-linking whenever the caps are non-fixed,
not just when there are multiple types - use gst_pad_is_fixed()
to test.
2007-08-27 11:59:56 +00:00
Jan Schmidt
fc50d2dc64 ext/alsa/Makefile.am: There is no GST_PLUGINS_BASE_LIBS defined.
Original commit message from CVS:
* ext/alsa/Makefile.am:
There is no GST_PLUGINS_BASE_LIBS defined.
* ext/alsa/gstalsa.c:
* ext/alsa/gstalsasink.c: (gst_alsasink_delay):
* ext/alsa/gstalsasrc.c: (gst_alsasrc_delay):
Add support for ALSA 24-bit formats.
snd_pcm_delay can return an error code, especially
during XRUNS. In that case, the best we can do is assume
delay = 0.
* gst/audioconvert/Makefile.am:
Add flags from -base before any more-remote dependencies.
2007-08-24 15:28:33 +00:00
Davyd
bad084b01e gst/volume/gstvolume.*: Add support for int32, int24 and int8 to the volume element.
Original commit message from CVS:
Based on a patch by: Davyd <davyd at madeley dot id dot au>
* gst/volume/gstvolume.c: (volume_choose_func),
(volume_update_real_volume), (gst_volume_set_volume),
(gst_volume_init), (volume_process_int32),
(volume_process_int32_clamp), (volume_process_int24),
(volume_process_int24_clamp), (volume_process_int16),
(volume_process_int16_clamp), (volume_process_int8),
(volume_process_int8_clamp), (volume_update_volume), (plugin_init):
* gst/volume/gstvolume.h:
Add support for int32, int24 and int8 to the volume element.
Fixes #445529.
2007-08-23 20:45:45 +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
Stefan Kost
64b4aedf97 gst/volume/gstvolume.c: Enable liboil for float and add more details about problems with int16.
Original commit message from CVS:
* gst/volume/gstvolume.c:
Enable liboil for float and add more details about problems with
int16.
2007-08-22 11:20:28 +00:00
Wim Taymans
5c59b5a2aa gst/playback/gstplaybasebin.c: Only post buffering messages when we are a stream.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (queue_threshold_reached):
Only post buffering messages when we are a stream.
2007-08-16 11:20:56 +00:00
Michael Smith
1b7a0df57e gst/audiorate/gstaudiorate.c: Debug output fixes.
Original commit message from CVS:
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
Debug output fixes.
* tests/check/elements/audiorate.c: (do_perfect_stream_test),
(GST_START_TEST):
Change the number of buffers used; 500 is too many and leads to
timeouts.
2007-08-10 13:55:44 +00:00
Tim-Philipp Müller
2c9bef0180 gst/: Printf format fixes (#465028).
Original commit message from CVS:
* gst/playback/gstqueue2.c:
* gst/videorate/gstvideorate.c:
Printf format fixes (#465028).
2007-08-10 10:08:05 +00:00
Michael Smith
9f9e76bc99 gst/audiorate/gstaudiorate.c: If we have a large (> 1 second) discontinuity, push a series of smaller buffers rather ...
Original commit message from CVS:
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
If we have a large (> 1 second) discontinuity, push a series of
smaller buffers rather than a single very large buffer. Avoids
unreasonably large single buffer allocations when encountering a
large gap.
* tests/check/elements/audiorate.c: (GST_START_TEST),
(audiorate_suite):
Add a test for this.
2007-08-09 15:44:02 +00:00
Josep Torra Valles
9730f452ee gst/playback/gstplaybasebin.c: Fixes: #465015
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (group_commit),
(queue_remove_probe), (queue_threshold_reached):
Patch by: Josep Torra Valles <josep@fluendo.com>
Fixes: #465015
Make sure we remove the check_queues buffer probe from the
correct queue to avoid racily going back to "buffering 99%" when
buffering is actually complete.
Also, fix the spelling of Josep's surname in the ChangeLog.
2007-08-09 12:06:43 +00:00
Josep Torre Valles
382b710277 Add connection-speed property. Fixes #464690.
Original commit message from CVS:
Patch by: Josep Torre Valles <josep@fluendo.com>
* docs/plugins/gst-plugins-base-plugins.args:
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
(gst_uri_decode_bin_init), (gst_uri_decode_bin_set_property),
(gst_uri_decode_bin_get_property), (gen_source_element):
Add connection-speed property. Fixes #464690.
2007-08-08 15:05:22 +00:00
Josep Torre Valles
5e5aa7b402 gst/playback/: Move connection-speed property from playbin to playbasebin so that we can also configure it in source ...
Original commit message from CVS:
Patch by: Josep Torre Valles <josep@fluendo.com>
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
(gst_play_base_bin_init), (queue_threshold_reached),
(gen_source_element), (setup_substreams),
(gst_play_base_bin_set_property), (gst_play_base_bin_get_property),
(gst_play_base_bin_get_streaminfo_value_array):
* gst/playback/gstplaybasebin.h:
* gst/playback/gstplaybin.c: (gst_play_bin_class_init),
(gst_play_bin_set_property), (gst_play_bin_get_property),
(gst_play_bin_handle_redirect_message):
Move connection-speed property from playbin to playbasebin so that we
can also configure it in source elements that have the connection-speed
property. Fixes #464028.
Add some debug info here and there.
2007-08-07 14:14:54 +00:00
Sebastian Dröge
5310373def gst/audiotestsrc/gstaudiotestsrc.c: Properly respond to conversion queries. Fixes #464079.
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_query):
Properly respond to conversion queries. Fixes #464079.
2007-08-06 16:42:22 +00:00
Sebastian Dröge
6f397125d1 gst/audiotestsrc/gstaudiotestsrc.*: Add float/double and int32 support to audiotestsrc. Fixes #460422.
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_init),
(gst_audio_test_src_src_fixate), (gst_audio_test_src_setcaps),
(gst_audio_test_src_init_sine_table),
(gst_audio_test_src_change_wave), (gst_audio_test_src_create):
* gst/audiotestsrc/gstaudiotestsrc.h:
Add float/double and int32 support to audiotestsrc. Fixes #460422.
Also set the default volume to the default value specified in the
GParamSpec.
2007-08-03 19:53:11 +00:00
Jens Granseuer
ef33f2fdc4 gst/audioconvert/gstaudioquantize.c: Fix C89 incompatibilities and spelling of explanations. Fixes #463215.
Original commit message from CVS:
Patch by: Jens Granseuer <jensgr at gmx dot net>
* gst/audioconvert/gstaudioquantize.c:
Fix C89 incompatibilities and spelling of explanations. Fixes #463215.
2007-08-03 19:40:14 +00:00
Dan Williams
ace9335ae3 gst/playback/gstplaybasebin.c: Don't return NULL when querying the stream info value array but instead return an empt...
Original commit message from CVS:
Patch by: Dan Williams <dcbw at redhat dot com>
* gst/playback/gstplaybasebin.c:
(gst_play_base_bin_get_streaminfo_value_array):
Don't return NULL when querying the stream info value array but instead
return an empty array. Fixes #459204.
2007-07-23 11:18:35 +00:00
Tim-Philipp Müller
2271ec928f gst/playback/gsturidecodebin.c: Init debug category before using it.
Original commit message from CVS:
* gst/playback/gsturidecodebin.c:
Init debug category before using it.
2007-07-23 10:41:18 +00:00
Wim Taymans
e59c110631 gst/videorate/gstvideorate.c: Use boilerplate.
Original commit message from CVS:
* gst/videorate/gstvideorate.c: (gst_video_rate_init),
(gst_video_rate_query):
Use boilerplate.
Add latency query, might not be perfect yet but already works a lot
better. Fixes #442557.
2007-07-13 18:12:19 +00:00
Jan Schmidt
b6ee0fa3d6 gst/ffmpegcolorspace/gstffmpegcodecmap.c: Fix the r_mask test for RGBA32 on little-endian.
Original commit message from CVS:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_caps_to_pixfmt):
Fix the r_mask test for RGBA32 on little-endian.
Fix a stupid typo that would have obviously broken
compilation on big-endian, if anyone was testing.
2007-07-13 15:52:02 +00:00
Wim Taymans
3bac564cc0 gst/videotestsrc/videotestsrc.*: Add alpha to the color struct.
Original commit message from CVS:
* gst/videotestsrc/videotestsrc.c: (paint_hline_AYUV),
(paint_hline_str4):
* gst/videotestsrc/videotestsrc.h:
Add alpha to the color struct.
Use a default alpha value of 255 instead of 128.
2007-07-12 15:02:43 +00:00
Wim Taymans
c03d6a8757 gst/playback/gstplaybasebin.c: Clear the dynamic pads counter when starting a new uri. This makes reusing playbin wor...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (no_more_pads_full),
(setup_source):
Clear the dynamic pads counter when starting a new uri. This makes
reusing playbin work again.
Fixes #454264.
2007-07-12 12:01:20 +00:00
Jan Schmidt
6fa26a44e3 gst/ffmpegcolorspace/: Add 2 new pixel formats - ABGR32 and ARGB32, which are reflections of the existing BGRA32 and ...
Original commit message from CVS:
* 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: (img_convert),
(img_get_alpha_info):
Add 2 new pixel formats - ABGR32 and ARGB32, which are reflections
of the existing BGRA32 and RGBA32 formats with the alpha at the other
end of the word. Partially fixes #451908
2007-07-06 11:40:45 +00:00
Wim Taymans
d42ca1fd83 gst/adder/gstadder.c: Make getcaps more robust by not using the proxycaps function. This makes sure that we don't end...
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_sink_getcaps),
(gst_adder_request_new_pad):
Make getcaps more robust by not using the proxycaps function. This makes
sure that we don't end up recursively calling getcaps upstream.
See #316248.
2007-07-03 11:52:47 +00:00
Wim Taymans
d4dfef2a0b gst/audioconvert/audioconvert.c: Include math.h to fix compilation.
Original commit message from CVS:
* gst/audioconvert/audioconvert.c:
Include math.h to fix compilation.
2007-06-29 17:21:18 +00:00
Jan Schmidt
cae46813ca gst/ffmpegcolorspace/gstffmpegcodecmap.c: Add a mapping for YUV format "IYU1", which is a 4:1:1 packed pixel format, ...
Original commit message from CVS:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt):
Add a mapping for YUV format "IYU1", which is a 4:1:1 packed pixel
format, as produced by some dc1394 cameras like the iSight.
See http://www.fourcc.org/yuv.php#IYU1
2007-06-29 14:47:42 +00:00
Sebastian Dröge
dbb857b93b gst/audioconvert/: Implement dithering and noise shaping in audioconvert. By default now
Original commit message from CVS:
* gst/audioconvert/Makefile.am:
* gst/audioconvert/audioconvert.c: (audio_convert_get_func_index),
(check_default), (audio_convert_prepare_context),
(audio_convert_clean_context), (audio_convert_convert):
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_dithering_get_type),
(gst_audio_convert_ns_get_type), (gst_audio_convert_class_init),
(gst_audio_convert_init), (gst_audio_convert_set_caps),
(gst_audio_convert_set_property), (gst_audio_convert_get_property):
* gst/audioconvert/gstaudioconvert.h:
* gst/audioconvert/gstaudioquantize.c:
(gst_audio_quantize_setup_noise_shaping),
(gst_audio_quantize_free_noise_shaping),
(gst_audio_quantize_setup_dither),
(gst_audio_quantize_free_dither),
(gst_audio_quantize_setup_quantize_func),
(gst_audio_quantize_setup), (gst_audio_quantize_free):
* gst/audioconvert/gstaudioquantize.h:
Implement dithering and noise shaping in audioconvert. By default now
TPDF dithering (and no noise shaping) will be used when converting
from a higher bit depth to 20 bit depth or smaller, otherwise
everything will be as it is now.
For the last audioconvert in a pipeline it would make sense to
use some kind of noise shaping, enabling it by default for all
conversions would give undesired results though. Fixes #360246.
* tests/check/elements/audioconvert.c: (setup_audioconvert),
(GST_START_TEST):
Adjust unit test for the new audioconvert.
2007-06-28 20:37:58 +00:00
Wim Taymans
8c05f2ebc9 gst/playback/gstqueue2.c: Use other metrics as well when estimating the buffer level.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (apply_segment), (update_buffering):
Use other metrics as well when estimating the buffer level.
2007-06-28 11:06:56 +00:00
Wim Taymans
aac5185f3e gst/playback/gstplaybasebin.c: Small debug improvement.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (make_decoder), (setup_source):
Small debug improvement.
* gst/playback/gstqueue2.c: (apply_segment), (update_buffering),
(plugin_init):
Tweak the rate estimation period.
When calculating the buffer filledness in rate estimation mode, don't
mix it with other metrics.
2007-06-28 10:21:19 +00:00
Wim Taymans
c198d8000c gst/playback/gstdecodebin2.c: When creating the groups, allow for a 5 second, unlimited buffers preroll phase after w...
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_group_new),
(gst_decode_group_expose), (gst_decode_group_free), (add_fakesink):
When creating the groups, allow for a 5 second, unlimited buffers
preroll phase after which we expose the group.
When the group is exposed, use a small number of buffers up to a 2
second limit. Also disconnect the overrun signal from multiqueue when we
exposed the group because it is not needed anymore.
2007-06-28 09:46:11 +00:00
Edward Hervey
fa877be84c ext/ogg/gstoggdemux.c: The chain should be freed if we error out here, else it will leak.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_find_chains):
The chain should be freed if we error out here, else it will leak.
* gst/playback/gstdecodebin.c: (disconnect_unlinked_signals),
(cleanup_decodebin):
Don't forget to *properly* remove the signals, else it will leak.
2007-06-23 14:44:07 +00:00
Wim Taymans
3b2762a5b2 gst/playback/gstdecodebin2.c: When handling a delayed-caps notification case, mark the group as dynamic so that the n...
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (analyze_new_pad):
When handling a delayed-caps notification case, mark
the group as dynamic so that the nbdynamic count is
incremented and decremented correctly. Fixes: #449156
Patch by: Wim Taymans <wim@fluendo.com>
2007-06-20 11:09:03 +00:00
David Schleef
c4c28a764a gst/playback/gstqueue2.c: Fix compile error from ignored return value.
Original commit message from CVS:
* gst/playback/gstqueue2.c:
Fix compile error from ignored return value.
2007-06-16 03:42:14 +00:00
Michael Smith
6077bc0124 gst/videoscale/vs_4tap.c: Update tmpbuf for all neccesary rows, not just one, as is required when downscaling.
Original commit message from CVS:
* gst/videoscale/vs_4tap.c: (vs_image_scale_4tap_Y):
Update tmpbuf for all neccesary rows, not just one, as is required
when downscaling.
Fixes #402076.
2007-06-15 15:23:36 +00:00
Edward Hervey
be1f78d2e2 gst/playback/gstqueue2.c: Fix build on MacOSX.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (gst_queue_create_read):
Fix build on MacOSX.
2007-06-13 18:20:57 +00:00
Wim Taymans
2e541b29d4 gst/playback/gstqueue2.c: Fix a division by zero when the max percent is <= 0. Fixes #446572. also update the bufferi...
Original commit message from CVS:
Patches by: Thiago Sousa Santos <thiagossantos at gmail dot com>
* gst/playback/gstqueue2.c: (update_buffering),
(gst_queue_locked_enqueue):
Fix a division by zero when the max percent is <= 0. Fixes #446572.
also update the buffering status when receiving events. Fixes #446551.
2007-06-12 08:38:06 +00:00
Thiago Sousa Santos
4d83551490 gst/playback/gstqueue2.c: Wait for preroll before attempting to forward a duration query upstream.
Original commit message from CVS:
Based on patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
* gst/playback/gstqueue2.c: (gst_queue_peer_query),
(gst_queue_handle_src_query):
Wait for preroll before attempting to forward a duration query upstream.
Fixes #445505.
2007-06-11 11:32:26 +00:00
Wim Taymans
919029d9c5 gst/playback/gstqueue2.c: Fix compilation.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (gst_queue_get_range):
Fix compilation.
2007-06-07 09:11:27 +00:00
Thiago Sousa Santos
658fbf5039 gst/playback/gstqueue2.c: Add pull based scheduling and fix some deadlocks. Fixes #444523.
Original commit message from CVS:
Patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
* gst/playback/gstqueue2.c: (gst_queue_init),
(gst_queue_handle_sink_event), (gst_queue_chain),
(gst_queue_get_range), (gst_queue_src_checkgetrange_function),
(gst_queue_sink_activate_push), (gst_queue_src_activate_push),
(gst_queue_src_activate_pull):
Add pull based scheduling and fix some deadlocks. Fixes #444523.
Does not yet completely work because duration queries upstream won't
block yet.
2007-06-06 13:36:26 +00:00
Wim Taymans
1a31080014 Some more fseeko checks.
Original commit message from CVS:
* configure.ac:
* gst/playback/gstqueue2.c: (gst_queue_create_read):
Some more fseeko checks.
2007-06-06 09:08:50 +00:00
Sven Arvidsson
0cffe4be7d gst/subparse/gstsubparse.*: Add support for SubViewer version 1 and 2 subtitles. Fixes #394061.
Original commit message from CVS:
Based on a patch by Sven Arvidsson <sa at whiz dot se>:
* gst/subparse/gstsubparse.c: (parse_subrip),
(subviewer_unescape_newlines), (parse_subviewer),
(gst_sub_parse_data_format_autodetect),
(gst_sub_parse_format_autodetect), (gst_subparse_type_find):
* gst/subparse/gstsubparse.h:
Add support for SubViewer version 1 and 2 subtitles. Fixes #394061.
* tests/check/elements/subparse.c: (GST_START_TEST),
(subparse_suite):
Add a unit test for both SubViewer formats.
2007-06-05 21:36:11 +00:00
Michael Smith
6499fcdc2e gst/audiotestsrc/gstaudiotestsrc.c: Don't overflow intermediate values when seeking to large time values in audiotest...
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_do_seek):
Don't overflow intermediate values when seeking to large time values
in audiotestsrc.
2007-06-05 17:08:04 +00:00
Wim Taymans
837d4b1bb9 gst/playback/gstqueue2.c: Include stdio to define fseeko.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (gst_queue_have_data),
(gst_queue_create_read), (gst_queue_read_item_from_file),
(gst_queue_open_temp_location_file), (gst_queue_locked_enqueue):
Include stdio to define fseeko.
2007-06-05 17:02:13 +00:00
Wim Taymans
d4bb17ab7a gst/playback/gsturidecodebin.c: Make sure we name srcpads uniquely even when using different internal decodebins.
Original commit message from CVS:
* gst/playback/gsturidecodebin.c: (no_more_pads_full),
(new_decoded_pad), (remove_pads), (make_decoder), (setup_source),
(gst_uri_decode_bin_change_state):
Make sure we name srcpads uniquely even when using different internal
decodebins.
Signal no-more-pads when no more dynamic elements exist.
Remove pads on cleanup.
2007-06-05 16:17:30 +00:00
Thiago Sousa Santos
73e8934af9 gst/playback/gstqueue2.c: Add support for filebased buffering. Fixes #441264.
Original commit message from CVS:
Based on patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
* gst/playback/gstqueue2.c: (gst_queue_class_init),
(gst_queue_init), (gst_queue_finalize),
(gst_queue_write_buffer_to_file), (gst_queue_have_data),
(gst_queue_create_read), (gst_queue_read_item_from_file),
(gst_queue_open_temp_location_file),
(gst_queue_close_temp_location_file), (gst_queue_locked_flush),
(gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
(gst_queue_is_empty), (gst_queue_is_filled),
(gst_queue_change_state), (gst_queue_set_temp_location),
(gst_queue_set_property):
Add support for filebased buffering. Fixes #441264.
2007-06-05 16:14:23 +00:00
Wim Taymans
3840b5a20f gst/playback/gstdecodebin2.c: Add support for delayed caps fixation when autoplugging.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_factory_filter),
(analyze_new_pad), (connect_pad), (expose_pad), (caps_notify_cb),
(caps_notify_group_cb), (gst_decode_group_new),
(gst_decode_group_free):
Add support for delayed caps fixation when autoplugging.
Optimize cases where a multiqueue is not needed/wanted, like right after
anything that is not a demuxer.
2007-06-05 16:05:19 +00:00
Wim Taymans
56e2a6b516 gst/tcp/gstmultifdsink.*: Add support for remuve_flush.
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_client_status_get_type),
(gst_multi_fd_sink_class_init), (gst_multi_fd_sink_add_full),
(gst_multi_fd_sink_remove_flush),
(gst_multi_fd_sink_remove_client_link),
(gst_multi_fd_sink_handle_client_write),
(gst_multi_fd_sink_handle_clients):
* gst/tcp/gstmultifdsink.h:
Add support for remuve_flush.
2007-06-05 16:00:33 +00:00
Wim Taymans
5deb6e096d gst/playback/gstplaybasebin.c: Stop buffering when the group is commited because the queues filled up.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (queue_overrun),
(no_more_pads_full):
Stop buffering when the group is commited because the queues filled up.
Fixes #442024.
2007-05-29 13:38:35 +00:00
Jan Schmidt
d9504cf065 gst/playback/gstplaybasebin.c: Handle unknown or invalid pads without crashing, as might occur if a media file like a...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (new_decoded_pad_full):
Handle unknown or invalid pads without crashing, as might occur if
a media file like an mp3 is specified as a subtitle file.
Fixes: #410039
2007-05-24 11:15:32 +00:00
Jan Schmidt
c446f911d4 gst/playback/gstplaybin.c: Block the subtitle bin output queue before ghosting it and linking, then unblock after. Th...
Original commit message from CVS:
* gst/playback/gstplaybin.c: (add_sink), (dummy_blocked_cb),
(setup_sinks):
Block the subtitle bin output queue before ghosting it and linking,
then unblock after. This avoids spurious not-linked errors caused
by the queue starting up (because it gets linked when it is ghosted).
Fixes: #350299
2007-05-24 10:19:54 +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
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
Wim Taymans
a18a10e81f gst/playback/gstdecodebin2.c: Make decodebin2 autoplug depayloaders too.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_factory_filter):
Make decodebin2 autoplug depayloaders too.
* gst/playback/gsturidecodebin.c: (source_new_pad):
Set the newly created decoder in a usable state when autoplugging a
dynamic source such as RTSP.
2007-05-17 16:27:32 +00:00
Tim-Philipp Müller
2cd5f527fe gst/playback/gststreaminfo.c: Ignore video-codec tag for audio streams and ignore audio-codec tags for video streams....
Original commit message from CVS:
* gst/playback/gststreaminfo.c: (cb_probe):
Ignore video-codec tag for audio streams and ignore audio-codec tags
for video streams. Should make codec name collection a bit more
robust against sloppy demuxers that send tag events containing both
tags down each pad.
2007-05-17 16:11:03 +00:00
Wim Taymans
d33939800d gst/playback/gstqueue2.c: Tweak the buffering thresholds a little.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (update_rates):
Tweak the buffering thresholds a little.
Update the buffer size with the previously calculate rate instead of
only when we calculate a new rate so that we get smoother buffering
updates.
* gst/playback/Makefile.am:
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_base_init),
(gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
(gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_property),
(gst_uri_decode_bin_get_property), (unknown_type),
(add_element_stream), (no_more_pads_full), (no_more_pads),
(source_no_more_pads), (new_decoded_pad), (array_has_value),
(gen_source_element), (has_all_raw_caps), (analyse_source),
(remove_decoders), (make_decoder), (remove_source),
(source_new_pad), (setup_source), (decoder_query_init),
(decoder_query_duration_fold), (decoder_query_duration_done),
(decoder_query_position_fold), (decoder_query_position_done),
(decoder_query_latency_fold), (decoder_query_latency_done),
(decoder_query_seeking_fold), (decoder_query_seeking_done),
(decoder_query_generic_fold), (gst_uri_decode_bin_query),
(gst_uri_decode_bin_change_state), (plugin_init):
New element that intergrates a source, optional buffering element and
decodebin.
2007-05-17 15:22:44 +00:00
Wim Taymans
fa972968b2 gst/playback/gstqueue2.c: fix build.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (gst_queue_get_type),
(gst_queue_class_init), (gst_queue_finalize), (update_time_level),
(apply_segment), (apply_buffer), (update_buffering),
(reset_rate_timer), (update_rates), (gst_queue_locked_flush),
(gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
(gst_queue_handle_sink_event), (gst_queue_is_filled),
(gst_queue_chain), (gst_queue_push_one), (gst_queue_loop),
(plugin_init):
fix build.
2007-05-17 13:36:11 +00:00
Wim Taymans
ae69903ca1 gst/playback/: On our way to playbin2 this is the new network queue that does buffering all by itself using high and ...
Original commit message from CVS:
* gst/playback/Makefile.am:
* gst/playback/gstqueue2.c: (gst_queue_get_type),
(gst_queue_class_init), (gst_queue_init), (gst_queue_finalize),
(gst_queue_getcaps), (gst_queue_bufferalloc),
(gst_queue_acceptcaps), (update_time_level), (apply_segment),
(apply_buffer), (update_buffering), (reset_rate_timer),
(update_rates), (gst_queue_locked_flush),
(gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
(gst_queue_handle_sink_event), (gst_queue_is_empty),
(gst_queue_is_filled), (gst_queue_chain), (gst_queue_push_one),
(gst_queue_loop), (gst_queue_handle_src_event),
(gst_queue_handle_src_query), (gst_queue_sink_activate_push),
(gst_queue_src_activate_push), (gst_queue_change_state),
(gst_queue_set_property), (gst_queue_get_property), (plugin_init):
On our way to playbin2 this is the new network queue that does buffering
all by itself using high and low watermarks. It can also measure up and
downstream bandwidth to optimally size the queue.
2007-05-17 11:57:44 +00:00
Michael Smith
ab76fa091a gst/: Use the segment->last_stop value to calculate the next timestamp to generate after a seek; not the segment->sta...
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_do_seek):
* gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_do_seek):
Use the segment->last_stop value to calculate the next timestamp to
generate after a seek; not the segment->start value.
2007-05-17 11:16:14 +00:00
David Schleef
c655a27ab4 gst/videotestsrc/videotestsrc.*: Add support for video/x-raw-bayer.
Original commit message from CVS:
* gst/videotestsrc/videotestsrc.c:
* gst/videotestsrc/videotestsrc.h:
Add support for video/x-raw-bayer.
2007-05-15 03:53:11 +00:00
Jan Schmidt
1e2c327792 gst/typefind/gsttypefindfunctions.c: Consolidate and re-work our mpeg system stream detection to probe more packets a...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mpeg_sys_is_valid_pack),
(mpeg_sys_is_valid_pes), (mpeg_sys_is_valid_sys),
(mpeg_sys_type_find), (mpeg_ts_type_find), (mpeg4_video_type_find),
(mpeg_video_type_find), (mpeg_video_stream_type_find),
(plugin_init):
Consolidate and re-work our mpeg system stream detection to probe
more packets and produce a higher confidence result. Fixes a
regression caused by lowering the typefind probability last year
- related to bug #397810. Remove the redundant MPEG-1 specific
typefind function, as the new one detects both MPEG-1 & MPEG-2
happily.
Also cleanup the MPEG elementary and MPEG-TS detection functions a
little.
Tested against my media test directory, with some improvements and
no regressions.
2007-05-11 17:33:43 +00:00
Wim Taymans
56f01bc0cb gst/playback/gstplaybasebin.c: Connect to the new queue "pushing" signal instead of the broken "running" one.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (fill_buffer), (check_queue),
(queue_out_of_data):
Connect to the new queue "pushing" signal instead of the broken
"running" one.
2007-05-10 15:28:13 +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
736a5c082f gst/adder/gstadder.c (gst_adder_src_event, gst_adder_collected, gst_adder_change_state): gst/adder/gstadder.h (bps, o...
Original commit message from CVS:
* gst/adder/gstadder.c (gst_adder_src_event, gst_adder_collected,
gst_adder_change_state):
* gst/adder/gstadder.h (bps, offset, collect_event, segment,
segment_pending, segment_position, segment_rate):
Handle playback-rate on adder.
2007-05-08 19:24:01 +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
Michael Smith
03e4592e41 gst/audiorate/gstaudiorate.c: If a buffer doesn't have a timestamp, assume it's contiguous with the previous buffer, ...
Original commit message from CVS:
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
If a buffer doesn't have a timestamp, assume it's contiguous with
the previous buffer, and synthesise timestamps appropriately.
2007-05-03 13:16:21 +00:00
Edward Hervey
25d28aae98 gst/videorate/gstvideorate.c: There is no sensible way to handle incoming buffers which don't have a valid timestamp....
Original commit message from CVS:
* gst/videorate/gstvideorate.c: (gst_video_rate_chain):
There is no sensible way to handle incoming buffers which don't have a
valid timestamp. We therefore discard them and wait for the next one.
2007-05-03 10:47:22 +00:00
Tim-Philipp Müller
997621c9b9 gst/playback/: Better error message for text files.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (type_found), (plugin_init):
* gst/playback/gstdecodebin2.c: (plugin_init):
Better error message for text files.
2007-05-01 18:45:36 +00:00
Julien Moutte
d299d1c063 ext/theora/theoradec.c: Calculate buffer duration correctly to generate a perfect stream (#433888).
Original commit message from CVS:
2007-04-27  Julien MOUTTE  <julien@moutte.net>

* ext/theora/theoradec.c: (_theora_granule_time),
(theora_dec_push_forward), (theora_handle_data_packet),
(theora_dec_decode_buffer): Calculate buffer duration correctly
to generate a perfect stream (#433888).
* gst/audioresample/gstaudioresample.c:
(audioresample_check_discont): Glib provides ABS.
2007-04-27 15:33:46 +00:00
Sebastian Dröge
84c824b952 gst/audioconvert/gstaudioconvert.c: Initalize the AudioConvertCtx with zeroes, otherwise it will contain pointers to ...
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init):
Initalize the AudioConvertCtx with zeroes, otherwise it will contain
pointers to random memory which are passed to g_free() when
audio_convert_prepare_context() is called the first time.
2007-04-24 18:58:25 +00:00
Dan Williams
37a334ddb7 gst/videorate/gstvideorate.c: Don't leak incoming buffer if gst_pad_push() returns a non-OK flow. Fixes #432755.
Original commit message from CVS:
Patch by: Dan Williams <dcbw redhat com>
* gst/videorate/gstvideorate.c: (gst_video_rate_chain):
Don't leak incoming buffer if gst_pad_push() returns a
non-OK flow. Fixes #432755.
* tests/check/elements/videorate.c: (GST_START_TEST),
(videorate_suite):
Unit test for the above by Yours Truly.
2007-04-24 15:00:07 +00:00
Stefan Kost
d24aff28b2 gst/adder/gstadder.c: Fix non-flushing segmented seeks, Fixes #340060 for me
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_setcaps), (gst_adder_src_event),
(gst_adder_sink_event), (gst_adder_collected):
Fix non-flushing segmented seeks, Fixes #340060 for me
2007-04-23 20:04:28 +00:00
Tim-Philipp Müller
97cff37e11 gst/audioresample/gstaudioresample.c: Make more functions static, just because we can.
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c:
Make more functions static, just because we can.
2007-04-21 14:14:24 +00:00
Stefan Kost
23a2a0e224 gst/subparse/: Use GST_DISABLE_XML here
Original commit message from CVS:
* gst/subparse/gstsubparse.c:
* gst/subparse/samiparse.c:
Use GST_DISABLE_XML here
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
(gst_xvimagesink_xwindow_new), (gst_xvimagesink_get_xv_support),
(gst_xvimagesink_buffer_alloc),
(gst_xvimagesink_navigation_send_event):
* sys/xvimage/xvimagesink.h:
Include stdlib.h when using atoi.
* tests/check/elements/playbin.c: (playbin_suite):
Use GST_DISABLE_REGISTRY here
2007-04-20 10:42:24 +00:00
Tim-Philipp Müller
1b4546e52f gst/app/Makefile.am: Fix CFLAGS and hopefully #430594.
Original commit message from CVS:
* gst/app/Makefile.am:
Fix CFLAGS and hopefully #430594.
2007-04-17 10:56:37 +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
Thomas Vander Stichele
66a6f3413c gst/gdp/gstgdppay.c (gst_gdp_pay_chain): tests/check/pipelines/streamheader.c (tag_event_probe_cb,
Original commit message from CVS:
* gst/gdp/gstgdppay.c (gst_gdp_pay_chain):
* tests/check/pipelines/streamheader.c (tag_event_probe_cb,
GST_START_TEST, buffer_probe_cb, GST_START_TEST):
Fix a bug where serialized IN_CAPS buffers needed to be set IN_CAPS.
2007-04-14 12:34:55 +00:00
Thomas Vander Stichele
7f7c429ab1 add debug
Original commit message from CVS:
add debug
2007-04-13 22:10:58 +00:00
Thomas Vander Stichele
9b7fcfdcae debug changes
Original commit message from CVS:
debug changes
2007-04-13 21:09:04 +00:00
Wim Taymans
807258cc03 gst/videorate/gstvideorate.c: Add some debug.
Original commit message from CVS:
* gst/videorate/gstvideorate.c: (gst_video_rate_setcaps),
(gst_video_rate_chain):
Add some debug.
* tests/check/elements/videorate.c: (GST_START_TEST),
(videorate_suite):
Added check for videorate changing caps handling. Closes #421834.
2007-04-12 15:00:03 +00:00
Thomas Vander Stichele
8a6b8cfb37 log tweaking
Original commit message from CVS:
log tweaking
2007-04-12 10:38:03 +00:00
Wim Taymans
3e455f8a5b gst/gdp/gstgdppay.c: Make sure we set the IN_CAPS flag correctly.
Original commit message from CVS:
* gst/gdp/gstgdppay.c: (gst_gdp_pay_chain),
(gst_gdp_pay_sink_event):
Make sure we set the IN_CAPS flag correctly.
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
Get the IN_CAPS flag before we call functions that mess with the flags.
2007-04-12 10:03:22 +00:00
Thomas Vander Stichele
bf82440579 gst/gdp/gstgdppay.c (gst_gdp_pay_reset_streamheader, gst_gdp_pay_chain, gst_gdp_pay_sink_event):
Original commit message from CVS:
* gst/gdp/gstgdppay.c (gst_gdp_pay_reset_streamheader,
gst_gdp_pay_chain, gst_gdp_pay_sink_event):
Only stamp buffers with offset/offset_end right before they get
pushed.  This ensures offset continuity, which was not the case
before as shown by
gst-launch -v -m audiotestsrc num-buffers=10 ! audioconvert ! vorbisenc ! gdppay ! identity check-imperfect-offset=TRUE ! fakesink silent=TRUE
2007-04-10 20:37:05 +00:00
Thomas Vander Stichele
5ce14bdb32 adding debugging
Original commit message from CVS:
adding debugging
2007-04-10 20:25:06 +00:00
Wim Taymans
34a49a9a06 gst/playback/gstplaybin.c: Activate sync in playbin, we are ready to handle it for live streams.
Original commit message from CVS:
* gst/playback/gstplaybin.c: (add_sink),
(gst_play_bin_change_state):
Activate sync in playbin, we are ready to handle it for live streams.
2007-04-06 12:58:06 +00:00
Tommi Myöhänen
32a727628f gst/playback/gstplaybasebin.c: Fix leak in add_stream(), when g_value_set_object() increases the refcount of streamin...
Original commit message from CVS:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* gst/playback/gstplaybasebin.c: (add_stream):
Fix leak in add_stream(), when g_value_set_object() increases the
refcount of streaminfo object. Fixes #426250.
2007-04-05 10:27:06 +00:00
David Schleef
e859791a21 gst/videotestsrc/: Add a test pattern called "circular", which has concentric rings with varying radial frequency. T...
Original commit message from CVS:
* gst/videotestsrc/gstvideotestsrc.c:
* gst/videotestsrc/gstvideotestsrc.h:
* gst/videotestsrc/videotestsrc.c:
* gst/videotestsrc/videotestsrc.h:
Add a test pattern called "circular", which has concentric
rings with varying radial frequency.  The main purpose of this
pattern is to test fidelity loss in a filter or scaler element.
Notably, this pattern is scale invariant, and is optimally viewed
with a width (and height) of 400.
2007-04-04 02:45:03 +00:00
Tommi Myöhänen
8676f3dce7 gst/playback/gstdecodebin2.c: Decodebin2 doesn't unref pads it obtains in some occasions:
Original commit message from CVS:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* gst/playback/gstdecodebin2.c: (connect_pad), (expose_pad),
(deactivate_free_recursive):
Decodebin2 doesn't unref pads it obtains in some occasions:
- multiqueue src pads, when either connecting further or exposing
- sink pads of new autoplugged elements
- peer pads when recursively freeing elements
Fixes #425455.
2007-04-03 11:10:52 +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
Andy Wingo
af17f81a47 gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Make perfect offsets also, not just timestamps.
Original commit message from CVS:
2007-03-29  Andy Wingo  <wingo@pobox.com>

* gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Make
perfect offsets also, not just timestamps.

* tests/check/elements/videorate.c (test_more): Test that given
any incoming offsets, that videorate produces perfect offsets.
2007-03-29 11:24:47 +00:00
Sebastian Dröge
293a9c09b8 gst/audioconvert/audioconvert.c: Add docs to the integer pack functions and implement proper rounding. Before we had ...
Original commit message from CVS:
* gst/audioconvert/audioconvert.c:
Add docs to the integer pack functions and implement proper
rounding. Before we had rounding towards negative infinity, i.e.
always the smaller number was taken. Now we use natural rounding,
i.e. rounding to the nearest integer and to the one with the largest
absolute value for X.5. The old rounding introduced some minor
distortions. Fixes #420079
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Fix one unit test that assumed the old rounding and added unit tests
for checking signed/unsigned int16 <-> signed/unsigned int16 with
depth 8, one for signed int16 <-> unsigned int16 and one for the new
rounding from signed int32 to signed/unsigned int16.
2007-03-27 12:44:14 +00:00
Michael Smith
e1544977a6 gst/audioconvert/gstaudioconvert.c: Fix typo in debug line introduced recently, as pointed out on irc.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (strip_width_64),
(gst_audio_convert_transform_caps):
Fix typo in debug line introduced recently, as pointed out on irc.
2007-03-27 11:31:17 +00:00
Michael Smith
b3827533a7 gst/videorate/gstvideorate.c: If videorate changes caps, we can no longer use the old buffer (which may have a differ...
Original commit message from CVS:
* gst/videorate/gstvideorate.c: (gst_video_rate_setcaps),
(gst_video_rate_reset), (gst_video_rate_chain):
If videorate changes caps, we can no longer use the old buffer
(which may have a different size, incompatible with our caps).
So don't do that; just duplicate the new frame more times.
2007-03-23 12:32:33 +00:00
Jan Schmidt
9cbead077e gst/playback/gstplaybin.c: Remove playbin's override of the set_clock vmethod. It's irrelevant after Wim's commit on ...
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gst_play_bin_class_init):
Remove playbin's override of the set_clock vmethod. It's irrelevant
after Wim's commit on the 19th.
2007-03-22 17:43:52 +00:00
Tim-Philipp Müller
5b1cd74011 gst/playback/gstplaybin.c: Use GST_PTR_FORMAT to log caps.
Original commit message from CVS:
* gst/playback/gstplaybin.c: (add_sink):
Use GST_PTR_FORMAT to log caps.
2007-03-21 11:03:23 +00:00
Young-Ho Cha
77cf4f207c gst/subparse/samiparse.c: Special-case some more colour names that pango doesn't handle by default. Fixes #420578.
Original commit message from CVS:
Patch by: Young-Ho Cha <ganadist at chollian net>
* gst/subparse/samiparse.c: (handle_start_font):
Special-case some more colour names that pango doesn't handle by
default. Fixes #420578.
2007-03-21 10:23:11 +00:00
Wim Taymans
d24780a03b gst/playback/gstdecodebin.c: Don't post STATE_DIRTY anymore.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (remove_fakesink):
Don't post STATE_DIRTY anymore.
* gst/playback/gstplaybin.c: (add_sink), (gst_play_bin_send_event),
(gst_play_bin_change_state):
Remove stream_time reset in seek handling, core does that now.
Disable clocking for live pipelines by forcing a NULL clock to the
complete pipeline, core is too smart now for our previous hack.
We can always autoplug in PAUSED now.
2007-03-19 10:52:50 +00:00
Michael Smith
3bc107dd77 gst/audioconvert/gstaudioconvert.c: Previous fix was too simplistic, and broke the tests. Use a better approach; only...
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
(strip_width_64), (append_with_other_format):
Previous fix was too simplistic, and broke the tests. Use a better
approach; only strip 64 from widths for integer audio.
2007-03-16 17:29:09 +00:00
Michael Smith
5759241eb4 gst/audioconvert/gstaudioconvert.c: We don't support 64 bit integer audio, so don't try to claim we can.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
(gst_audio_convert_transform_caps):
We don't support 64 bit integer audio, so don't try to claim we can.
Stops us producing caps don't match our template caps.
Update comments.
2007-03-16 16:42:23 +00:00
Michael Smith
4ab2d699fd gst/audioresample/gstaudioresample.c: Don't trigger discontinuities for very small imperfections; a filter flush will...
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c:
(audioresample_check_discont), (audioresample_transform):
Don't trigger discontinuities for very small imperfections; a filter
flush will sound bad, and many plugins have rounding errors leading
to these.
2007-03-15 10:52:21 +00:00
Julien Moutte
6940042ecf gst/audioresample/gstaudioresample.c: Handle discontinuous streams.
Original commit message from CVS:
2007-03-14  Julien MOUTTE  <julien@moutte.net>

* gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
(audioresample_transform_size), (audioresample_do_output),
(audioresample_transform), (audioresample_pushthrough): Handle
discontinuous streams.
* gst/audioresample/gstaudioresample.h:
* tests/check/elements/audioresample.c:
(test_discont_stream_instance), (GST_START_TEST),
(audioresample_suite): Add a test for discontinuous streams.
* win32/common/config.h: Updated.
2007-03-14 17:16:30 +00:00
Thomas Vander Stichele
9fe1aa7a1a add buffer logging
Original commit message from CVS:
add buffer logging
2007-03-14 15:05:32 +00:00
Thomas Vander Stichele
081deac039 gst/audioresample/: Since I really am not interested in a debug line for each sample being processed, move the librar...
Original commit message from CVS:
* gst/audioresample/debug.h:
* gst/audioresample/resample.c: (resample_init):
Since I really am not interested in a debug line for each sample
being processed, move the library's debugging to its own category,
libaudioresample
2007-03-14 14:48:12 +00:00
Thomas Vander Stichele
1587ea7bba add debugging and reformat docs
Original commit message from CVS:
add debugging and reformat docs
2007-03-14 14:09:21 +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
Kamil Pawlowski
389eb6f2f8 gst/subparse/: Add support for MPL2 subtitle format (#413799).
Original commit message from CVS:
Patch by: Kamil Pawlowski  <kamilpe gmail com>
* gst/subparse/Makefile.am:
* gst/subparse/gstsubparse.c:
(gst_sub_parse_data_format_autodetect),
(gst_sub_parse_format_autodetect), (gst_sub_parse_sink_event),
(gst_subparse_type_find):
* gst/subparse/gstsubparse.h:
* gst/subparse/mpl2parse.c: (mpl2_parse_line), (parse_mpl2):
* gst/subparse/mpl2parse.h:
Add support for MPL2 subtitle format (#413799).
2007-03-10 11:17:52 +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
b9ca5573a9 ext/pango/gsttextoverlay.c: Some more logging. Only accept newsegment events in TIME format and send a WARNING messag...
Original commit message from CVS:
* ext/pango/gsttextoverlay.c: (gst_text_overlay_text_event),
(gst_text_overlay_video_event):
Some more logging. Only accept newsegment events in TIME format and
send a WARNING message if they are not in TIME format.
* gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
(gst_sub_parse_init), (gst_sub_parse_src_event), (handle_buffer),
(gst_sub_parse_chain), (gst_sub_parse_sink_event):
* gst/subparse/gstsubparse.h:
No need to allocate GstSegment structure dynamically, just put it
into the instance structure; ignore newsegment events in BYTE
format and in particular don't let it overwrite our saved TIME
segment from the last seek.
2007-03-09 16:28:04 +00:00
Michael Smith
c9cd89bbab gst/typefind/gsttypefindfunctions.c: Replace AC3 typefinder with one that isn't terrible, and actually works usefully.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (ac3_type_find):
Replace AC3 typefinder with one that isn't terrible, and actually
works usefully.
2007-03-09 13:05:04 +00:00
Thomas Vander Stichele
e81b9ec719 gst/audioconvert/gstaudioconvert.c: fix error category and translatable string
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_transform):
fix error category and translatable string
2007-03-09 12:22:53 +00:00
Tim-Philipp Müller
3eba8b10e2 gst/subparse/gstsubparse.c: Break out of loop in chain function as soon as possible if we get a non-OK flow return.
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (handle_buffer):
Break out of loop in chain function as soon as possible if we get
a non-OK flow return.
2007-03-09 10:49:53 +00:00
Jan Schmidt
46a7bd8dfd ext/alsa/gstalsasink.c: Extra log line.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_open):
Extra log line.
* ext/pango/gstclockoverlay.c: (gst_clock_overlay_init):
* ext/pango/gsttimeoverlay.c: (gst_time_overlay_init):
Use pango_font_description_set_family_static instead of
pango_font_description_set_family to save a string copy (it was
leaking due to the strdup anyway)
* gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_finalize):
* gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_finalize):
* gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_finalize):
* gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_finalize):
Chain up in finalize.
2007-03-08 12:53:51 +00:00
Thomas Vander Stichele
9672eea8b4 update docs
Original commit message from CVS:
update docs
2007-03-07 17:05:21 +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
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
Jan Schmidt
c73e88a6a1 Add documentation for decodebin2 that indicates that the API is still unstable.
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:
* docs/plugins/inspect/plugin-decodebin2.xml:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
Add documentation for decodebin2 that indicates that the API
is still unstable.
2007-03-02 12:59:15 +00:00
Ed Catmur
61eaa5eb9b gst/playback/gstplaybin.c: Fix race condition when rapidly switching visualisations in playbin.
Original commit message from CVS:
Patch by: Ed Catmur <ed at catmur dot co dot uk>
* gst/playback/gstplaybin.c: (gst_play_bin_vis_unblocked),
(gst_play_bin_vis_blocked), (gst_play_bin_set_property):
Fix race condition when rapidly switching visualisations in playbin.
Fixes #401029.
2007-02-28 15:17:20 +00:00
Stefan Kost
14cd6e6ae1 gst/audioconvert/audioconvert.c: Also make valgrind happy and avoid copying data in some cases.
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
(double_hq), (audio_convert_get_func_index), (check_default),
(audio_convert_prepare_context), (audio_convert_convert):
Also make valgrind happy and avoid copying data in some cases.
2007-02-28 12:08:27 +00:00
Stefan Kost
2c67c89457 Don't run inplace if that overwrites source data as we go. Add more tests. Fixes #339837 even more.
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
(double_hq), (audio_convert_get_func_index),
(audio_convert_prepare_context), (audio_convert_convert):
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_class_init), (gst_audio_convert_get_unit_size),
(gst_audio_convert_transform_caps):
* tests/check/elements/audioconvert.c: (GST_START_TEST),
(audioconvert_suite):
Don't run inplace if that overwrites source data as we go. Add more
tests. Fixes #339837 even more.
2007-02-28 11:47:45 +00:00
David Schleef
749d720436 Add a new plugin/library to make it easy for apps to shove data into a pipeline.
Original commit message from CVS:
* configure.ac:
* gst/app/Makefile.am:
* gst/app/gstapp.c:
* gst/app/gstappsrc.c:
* gst/app/gstappsrc.h:
Add a new plugin/library to make it easy for apps to shove
data into a pipeline.
2007-02-26 21:01:03 +00:00
Loïc Minier
2770d0fbea Fix compilation with LDFLAGS='-Wl,-z,defs' (#410963).
Original commit message from CVS:
Patch by: Loïc Minier <lool+gnome at via ecp fr>
* configure.ac:
* ext/alsa/Makefile.am:
* gst/audiotestsrc/Makefile.am:
Fix compilation with LDFLAGS='-Wl,-z,defs' (#410963).
2007-02-24 20:12:49 +00:00
Tim-Philipp Müller
d35f4efe67 gst/playback/gstplaybin.c: Improve docs: point out that the application needs to assist playbin with buffering.
Original commit message from CVS:
* gst/playback/gstplaybin.c:
Improve docs: point out that the application needs to assist playbin
with buffering.
2007-02-23 18:49:29 +00:00
Stefan Kost
00d7c52de8 Add float as an intermediate format, as well as float mixing. Enable test that was failing before. Fixes #339837
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
(double_hq), (audio_convert_get_func_index),
(audio_convert_prepare_context), (audio_convert_convert):
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_setup_matrix),
(gst_channel_mix_mix_int), (gst_channel_mix_mix_float):
* gst/audioconvert/gstchannelmix.h:
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Add float as an intermediate format, as well as float mixing. Enable
test that was failing before. Fixes #339837
2007-02-22 09:04:37 +00:00
Stefan Kost
1b01b41b69 gst/volume/gstvolume.c: Unbreak volume, value remains gint.
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_process_int16),
(volume_process_int16_clamp), (volume_set_caps):
Unbreak volume, value remains gint.
2007-02-21 13:55:54 +00:00
Stefan Kost
b8dfedde6a gst/volume/gstvolume.*: Extend float audio support (double) and some int->uint cleanups.
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_choose_func),
(volume_update_real_volume), (gst_volume_set_volume),
(gst_volume_init), (volume_process_double), (volume_process_float),
(volume_process_int16), (volume_process_int16_clamp),
(volume_set_caps), (volume_transform_ip), (volume_update_volume):
* gst/volume/gstvolume.h:
Extend float audio support (double) and some int->uint cleanups.
2007-02-21 13:08:51 +00:00
Edward Hervey
7bc6fc07a1 gst/playback/gstdecodebin2.c: Don't free groups from the streaming threads. Just put them aside and free them in disp...
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_dispose),
(multi_queue_underrun_cb), (gst_decode_group_check_if_drained),
(sort_end_pads), (gst_decode_group_expose),
(gst_decode_group_hide):
Don't free groups from the streaming threads. Just put them aside and
free them in dispose.
2007-02-20 15:44:32 +00:00
Edward Hervey
7746c2f176 gst/playback/gstdecodebin2.c: Handle dynamic pads within groups.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (connect_element),
(pad_added_group_cb), (gst_decode_group_check_if_blocked),
(sort_end_pads), (gst_decode_group_expose):
Handle dynamic pads within groups.
Sort pads before exposing them in order to make playbin happy.
There still is a race with the multiqueue filling up. This should be
solved separately.
Fixes #398721
2007-02-20 11:20:52 +00:00
Yves Lefebvre
a6e3e71fe3 gst/videorate/gstvideorate.c: Don't leak caps. Fixes #408278.
Original commit message from CVS:
Patch by: Yves Lefebvre  <ivanohe abacom com>
* gst/videorate/gstvideorate.c: (gst_video_rate_setcaps):
Don't leak caps. Fixes #408278.
2007-02-16 10:15:46 +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
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
Tim-Philipp Müller
a58a6b1854 gst/playback/gstplaybasebin.c: When we have external subtitles and wait for the subtitle decodebin to get up and runn...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (setup_source):
When we have external subtitles and wait for the subtitle decodebin
to get up and running, we set up a (sync) bus handler for the
subtitle decodebin, so we can stop waiting when it posts an error
message. However, we should do that before we set the subtitle
decodebin's state to playing, otherwise things are racy and we might
miss error messages posted before we had a chance to set up the bus.
This should finally fix totem hanging on .txt pseudo-subtitle files.
2007-02-12 10:33:40 +00:00
Tim-Philipp Müller
5f186351a1 gst/subparse/gstsubparse.c: For SubRip (.srt) subtitles, ignore all markup tags we don't handle (like font tags, for ...
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (subrip_remove_unhandled_tag),
(subrip_remove_unhandled_tags), (parse_subrip):
For SubRip (.srt) subtitles, ignore all markup tags we don't
handle (like font tags, for example).
* tests/check/elements/subparse.c:
Add test for this.
2007-02-10 18:19:37 +00:00
Tim-Philipp Müller
12f5dd47ef gst/playback/: Don't error out if there is no fakesink in the READY to NULL state change, since when decodebin is re-...
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (add_fakesink),
(gst_decode_bin_change_state):
* gst/playback/gstdecodebin2.c: (add_fakesink),
(gst_decode_bin_change_state):
Don't error out if there is no fakesink in the READY to NULL state
change, since when decodebin is re-used, we're only adding the
fakesink element in READY to PAUSED.
* tests/check/elements/decodebin.c:
(new_decoded_pad_plug_fakesink_cb), (GST_START_TEST),
(decodebin_suite):
Minimal unit test to make sure we can use the same decodebin
instance twice (at least with audiotestsrc input).
2007-02-09 13:16:27 +00:00
Tim-Philipp Müller
b827e4d539 gst/audioconvert/gstaudioconvert.c: Fix up docs chunk so that gtk-doc doesn't complain, and fix typo.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
Fix up docs chunk so that gtk-doc doesn't complain, and fix typo.
2007-02-07 13:05:01 +00:00
Stefan Kost
66727fe484 gst/audioconvert/gstaudioconvert.c: Also mention that a conversion from double to float is suboptimal still.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
Also mention that a conversion from double to float is suboptimal still.
2007-02-06 14:00:31 +00:00
Tim-Philipp Müller
2933302ce8 gst/videoscale/gstvideoscale.c: A width and height of 1 makes us crash, so increase minimum size to 2x2 pixels until ...
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c:
A width and height of 1 makes us crash, so increase minimum size to
2x2 pixels until someone feels like fixing this (#404512).
2007-02-05 11:44:52 +00:00
Tim-Philipp Müller
b7cf10eb6d gst/audioconvert/gstaudioconvert.c: We don't support floats with a width of 40, 48 or 56 bits.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(set_structure_widths_32_and_64), (make_lossless_changes):
We don't support floats with a width of 40, 48 or 56 bits.
2007-02-02 11:21:48 +00:00
Stefan Kost
36180a5f3d gst/audioconvert/: Support for 64-bit float audio in audioconvert (#339837)
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (float), (double),
(audio_convert_get_func_index):
* gst/audioconvert/gstaudioconvert.c: (set_structure_widths),
(make_lossless_changes):
Support for 64-bit float audio in audioconvert (#339837)
2007-02-02 09:48:53 +00:00
Jan Schmidt
800cd83487 gst/playback/gstplaybin.c: Add audioresample+audioconvert in front of the visualisation element, so that elements lik...
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gen_vis_element):
Add audioresample+audioconvert in front of the visualisation
element, so that elements like libvisual 0.4 that don't support all
samplerates can work.
Fixes: #402505
2007-01-31 15:58:53 +00:00
Tim-Philipp Müller
514c2d7610 gst/playback/gstplaybasebin.c: Take some locks and make a copy of the streaminfo value array we maintain while holdin...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_property),
(gst_play_base_bin_get_streaminfo_value_array):
Take some locks and make a copy of the streaminfo value array we
maintain while holding the lock, so that the application can
retrieve the stream-info as a value array in a thread-safe way.
2007-01-30 19:19:37 +00:00
Wim Taymans
75d5fcb62e gst/audioconvert/gstaudioconvert.c: Don't fail on 0 sized buffers. Fixes #396835.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
Don't fail on 0 sized buffers. Fixes #396835.
2007-01-30 11:29:17 +00:00
David Schleef
888ea4730e gst/typefind/gsttypefindfunctions.c: Detect BBCD as video/x-dirac, so we can play raw dirac streams.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c:
Detect BBCD as video/x-dirac, so we can play raw dirac
streams.
2007-01-29 21:13:07 +00:00
Wim Taymans
ce1ff3c94c gst/playback/gstplaybasebin.c: Don't try to disconnect a signal from a finalized object.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (remove_source):
Don't try to disconnect a signal from a finalized object.
2007-01-25 16:02:41 +00:00
Tim-Philipp Müller
af52900954 gst/playback/gstdecodebin2.c: Cast lock macro parameters to make sure we're actually accessing the lock member at the...
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_dispose):
Cast lock macro parameters to make sure we're actually accessing the
lock member at the right class level. Free list itself in _dispose()
as well and NULL it in case dispose gets called multiple times.
2007-01-25 14:29:21 +00:00
Edward Hervey
a8d29e1f82 gst/playback/gstdecodebin2.c: Free GstDecodeGroups no longer used.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c:
(gst_decode_bin_dispose),(gst_decode_bin_finalize):
Free GstDecodeGroups no longer used.
(gst_decode_group_expose):
Don't unlock too many times !
(deactivate_free_recursive):
Free iterator once we're done with it.
Fix for recursively deactivating elements (stop at ghostpads).
2007-01-25 14:02:37 +00:00
Tim-Philipp Müller
dab1309f32 gst/playback/gstplaybin.c: Fix up caps on the frame buffer before we save it and potentially make it accessible to ot...
Original commit message from CVS:
* gst/playback/gstplaybin.c: (handoff):
Fix up caps on the frame buffer before we save it and potentially
make it accessible to other threads via g_object_get; also use
gst_buffer_replace() instead of gst_mini_object_replace().
2007-01-25 12:24:18 +00:00
Tim-Philipp Müller
d289e2d55c gst/playback/gstplaybin.c: Make getting the current frame thread-safe.
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gst_play_bin_get_property):
Make getting the current frame thread-safe.
2007-01-25 12:06:59 +00:00
Edward Hervey
77cf01b76b gst/playback/gstdecodebin2.c: Set queues to bigger sizes to cope with HD contents.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_finalize),
(gst_decode_group_new), (gst_decode_group_free):
Set queues to bigger sizes to cope with HD contents.
Fix some mutex freeing and add comment about MT safe methods.
2007-01-25 11:48:10 +00:00
Tim-Philipp Müller
b8d04c8843 gst/playback/gstdecodebin2.c: Don't leak mutex.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_finalize):
Don't leak mutex.
* tests/check/elements/playbin.c:
(test_sink_usage_video_only_stream),
(test_suburi_error_unknowntype), (test_suburi_error_invalidfile),
(test_suburi_error_wrongproto), (test_missing_urisource_handler),
(test_missing_suburisource_handler),
(test_missing_primary_decoder), (playbin_suite):
Run all tests once with decodebin and once with decodebin2.
One test does not pass yet with decodebin2.
2007-01-23 17:49:29 +00:00
Edward Hervey
a5146c9037 gst/playback/gstdecodebin2.c: Cleanups.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_group_new),
(get_current_group), (group_demuxer_event_probe),
(gst_decode_group_expose), (deactivate_free_recursive),
(gst_decode_group_free):
Cleanups.
Don't forget to emit 'no-more-pads' once a group is exposed.
Cleanup elements from a DecodeGroup once we remove it.
Protect call to gst_decode_group_expose() with the decodebin lock.
2007-01-22 17:37:38 +00:00
Tim-Philipp Müller
0da3960238 gst/typefind/gsttypefindfunctions.c: Don't go into an endless loop if the file starts with 00 00 01 2X, like quicktim...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mpeg4_video_type_find):
Don't go into an endless loop if the file starts with 00 00 01 2X,
like quicktime redirect files might. Fixes #396042.
* tests/check/Makefile.am:
* tests/check/gst/.cvsignore:
* tests/check/gst/typefindfunctions.c: (GST_START_TEST),
(typefindfunctions_suite):
Add unit test for the above.
2007-01-22 12:03:27 +00:00
Tim-Philipp Müller
acf3bcdfbb gst/playback/gstplaybasebin.c: Attempt at a better error message in case we don't have the required
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (setup_subtitle),
(gen_source_element), (gst_play_base_bin_change_state):
Attempt at a better error message in case we don't have the required
URI handler installed; post missing-plugin message also when we're
missing an URI handler for the subtitle URI; clean up properly also
when an error occurs and we never made it to PAUSED state.
* tests/check/elements/playbin.c: (GST_START_TEST),
(playbin_suite):
Check that we're also getting a missing-plugin messsage for a
missing subtitle URI handler (and clean up properly).
2007-01-19 19:09:05 +00:00
Tim-Philipp Müller
638dbd7b71 gst/playback/gstplaybasebin.c: Plug a few reference leaks.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (analyse_source), (setup_source):
Plug a few reference leaks.
2007-01-19 18:47:30 +00:00
Tim-Philipp Müller
91773593cf gst/typefind/gsttypefindfunctions.c: Lower probability a bit if the marker isn't right at the start, to decrease the ...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
Lower probability a bit if the marker isn't right at the start,
to decrease the chance of false positives.
2007-01-19 12:23:06 +00:00
Tim-Philipp Müller
c47074371a gst/typefind/gsttypefindfunctions.c: Small mpeg2 system stream typefinding improvement: make typefinder probe a bit i...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
Small mpeg2 system stream typefinding improvement: make typefinder
probe a bit into the stream instead of just looking for a marker
at the beginning. Fixes #397810.
2007-01-19 11:31:50 +00:00
Tim-Philipp Müller
4936d6ba02 gst/audioconvert/gstchannelmix.c: Remove compatibility cruft for prehistoric GLib versions.
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c:
Remove compatibility cruft for prehistoric GLib versions.
2007-01-18 16:23:35 +00:00
Tim-Philipp Müller
5262208d6e gst/playback/: Let decodebin be the element to post missing-plugin messages for missing decoders (rather than playbin...
Original commit message from CVS:
* gst/playback/Makefile.am:
* gst/playback/gstdecodebin.c: (close_pad_link):
* gst/playback/gstdecodebin2.c: (analyze_new_pad):
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
(gst_play_base_bin_handle_message_func), (unknown_type):
Let decodebin be the element to post missing-plugin messages for
missing decoders (rather than playbin); make playbin implement
GstBin::handle_message so we can suppress missing-plugin messages
for types we're not handling on purpose (don't want to bring up an
installer in those cases).
2007-01-17 16:11:14 +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
Tim-Philipp Müller
69a3ef4f92 gst/typefind/gsttypefindfunctions.c: Fix comment.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c:
Fix comment.
2007-01-12 09:45:23 +00:00
Tim-Philipp Müller
9de43e8bc5 gst/playback/gstplaybin.c: Post missing-plugin messages also when we error out because converters, textoverlay or aut...
Original commit message from CVS:
* gst/playback/gstplaybin.c: (post_missing_element_message),
(gen_video_element), (gen_text_element), (gen_audio_element),
(gen_vis_element):
Post missing-plugin messages also when we error out because
converters, textoverlay or auto*sinks are missing (#161922).
2007-01-11 13:12:17 +00:00
Wim Taymans
3f147830e1 gst/playback/: Fix the case where we try to ref a NULL element when we delay a link because of unfixed caps.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (dynamic_add), (close_pad_link),
(is_demuxer_element), (new_caps):
* gst/playback/gstplaybasebin.c: (source_new_pad):
Fix the case where we try to ref a NULL element when we delay a link
because of unfixed caps.
Set the state of autoplugged decodebins to PAUSED.
RTSP now works in playbin, we can remove it from the blacklist.
2007-01-10 16:08:18 +00:00
Tim-Philipp Müller
686ad73bfe gst/playback/: Post missing-plugin messages on the bus for missing sources and missing decoders/demuxers/depayloaders...
Original commit message from CVS:
* gst/playback/Makefile.am:
* gst/playback/gstplaybasebin.c: (string_arr_has_str),
(unknown_type), (setup_subtitle), (gen_source_element):
* gst/playback/gstplaybin.c: (plugin_init):
Post missing-plugin messages on the bus for missing sources and
missing decoders/demuxers/depayloaders; fix error code used when
we're missing an URI handler source; for media types that we are not
handling on purpose at the moment, don't print "don't know how to
handle xyz" messages to the terminal or post missing-plugin
messages on the bus.
* tests/check/elements/playbin.c: (create_playbin),
(GST_START_TEST), (gst_codec_src_uri_get_type),
(gst_codec_src_uri_get_protocols), (gst_codec_src_uri_get_uri),
(gst_codec_src_uri_set_uri), (gst_codec_src_uri_handler_init),
(gst_codec_src_init_type), (gst_codec_src_base_init),
(gst_codec_src_create), (gst_codec_src_class_init),
(gst_codec_src_init), (plugin_init), (playbin_suite):
Add some tests for the missing-plugin stuff.
2007-01-09 14:33:24 +00:00
Günter Thelen
7ccf512bce gst/typefind/gsttypefindfunctions.c: Add typefinder for flac-in-ogg in conformance with the ogg-mapping on flac.sf.ne...
Original commit message from CVS:
Patch by: Günter Thelen  <daedalus dot inc at gmx net>
* gst/typefind/gsttypefindfunctions.c: (flac_type_find),
(plugin_init):
Add typefinder for flac-in-ogg in conformance with the ogg-mapping
on flac.sf.net (there appear to be other versions of the first
ogg page in the wild) (#391365).
2007-01-08 14:01:23 +00:00
Tim-Philipp Müller
9052dc4681 gst/subparse/gstsubparse.*: Remove spurious 1000 subtrahend when calculating the timestamp from the frame number and ...
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (parse_mdvdsub):
* gst/subparse/gstsubparse.h:
Remove spurious 1000 subtrahend when calculating the timestamp from
the frame number and the frame rate . Also, use the frames/second
value specified in the first line of the file, if one is specified
there. Should fix #357503.
* tests/check/elements/subparse.c: (do_test),
(test_tmplayer_do_test), (test_microdvd_do_test), (GST_START_TEST),
(subparse_suite):
Add some basic unit tests for the microdvd subtitle format.
2007-01-08 12:30:03 +00:00
Tim-Philipp Müller
4e3fc9611f Printf format and missing argument fixes.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_stream_out):
* ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
* gst/playback/gstdecodebin2.c:
(gst_decode_group_check_if_blocked):
Printf format and missing argument fixes.
2007-01-05 19:43:55 +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
Jan Schmidt
3da0606ffc gst/ffmpegcolorspace/: Add 2 new caps arrangements, for 24-bit RGB and BGR in 32-bits, but at the other end of the wo...
Original commit message from CVS:
* 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: (img_convert),
(img_get_alpha_info):
Add 2 new caps arrangements, for 24-bit RGB and BGR in 32-bits, but at the
other end of the word. Fixes: #387073.
Add some inconsequential branch hints in a couple of places.
2006-12-22 11:09:34 +00:00
Tim-Philipp Müller
d07b7ac539 gst/ffmpegcolorspace/gstffmpegcodecmap.c: The "signed" field in raw audio caps is of boolean type, trying to extract ...
Original commit message from CVS:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_caps_to_smpfmt):
The "signed" field in raw audio caps is of boolean type, trying to
extract the value with _get_int() will fail (fix to keep in sync with
the copy in gst-ffmpeg)
2006-12-21 12:30:11 +00:00
Tim-Philipp Müller
a9c59c04e7 gst/typefind/gsttypefindfunctions.c: Forgot to register the extensions.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
Forgot to register the extensions.
2006-12-20 10:29:58 +00:00
Tim-Philipp Müller
2a18fb3546 gst/typefind/gsttypefindfunctions.c: Add typefinder for VIVO files (my christmas present to the 90s).
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (vivo_type_find),
(plugin_init):
Add typefinder for VIVO files (my christmas present to the 90s).
2006-12-20 09:25:55 +00:00
Tim-Philipp Müller
a4c0d0b8c3 gst/playback/gstdecodebin.c: Special-case the text/plain media type: we only want to recognise it as a 'raw' decoded ...
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (type_found):
Special-case the text/plain media type: we only want to recognise it
as a 'raw' decoded media type if it comes from a demuxer or subtitle
parser, but not if the entire stream is of text/plain type. If the
entire stream is text/plain, we should just error out.
This fixes playback of audio files with lyrics in totem. Totem can't
distinguish between text files and subtitle files and passes any
.txt file with the same basename as the main file to playbin as
suburi, and playbin will then throw a 'subtitle found, but no video
stream' error, which isn't entirely helpful. See #380342.
Also, with this change we'll show a slightly more correct error
message in case totem passes a playlist file to us (although a
custom error message wording instead of the default text would
probably not be a bad idea either).
Same problem also needs to be fixed for playbin+decodebin2.
* tests/check/Makefile.am:
* tests/check/elements/decodebin.c: (src_handoff_cb),
(decodebin_new_decoded_pad_cb), (GST_START_TEST),
(decodebin_suite):
Add simple unit test for decodebin for the above.
2006-12-16 13:59:09 +00:00
Tim-Philipp Müller
95f46c1fee gst/playback/: Refuse to change state to READY when we failed to create any of the required elements in our instance ...
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (gst_decode_bin_change_state):
* gst/playback/gstdecodebin2.c: (gst_decode_bin_change_state):
Refuse to change state to READY when we failed to create any of the
required elements in our instance init function.
2006-12-16 12:22:57 +00:00
Jan Schmidt
35fa7a7fbd gst/playback/gstdecodebin.c: Handle the case where an element has multiple pads with unfixed caps as well as still po...
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (find_dynamic), (dynamic_add),
(close_pad_link), (elem_is_dynamic), (unlinked), (close_link):
Handle the case where an element has multiple pads with
unfixed caps as well as still possibly producing more dynamic
pads by storing each case as a distinct entry in the dynamic list.
Fixes #38223 again.
2006-12-05 12:44:00 +00:00
Wim Taymans
bdd8747c6c gst/playback/gstdecodebin.c: Fix #382223, add more dynamic caps handling.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (close_pad_link):
Fix #382223, add more dynamic caps handling.
2006-12-04 13:02:37 +00:00
Michael Smith
ff480c70d4 gst/audiorate/gstaudiorate.c: Delete bad debug code.
Original commit message from CVS:
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
Delete bad debug code.
Fixes #381219
2006-12-01 10:36:50 +00:00
Sergey Scobich
748960f9ee Fix compilation on win32 under VS8
Original commit message from CVS:
* gst/videoscale/vs_4tap.c:
* win32/MANIFEST:
* win32/common/config.h:
* win32/vs8/libgstvideoscale.vcproj:
Fix compilation on win32 under VS8
Patch by: Sergey Scobich <sergey dot scobich at gmail dot com>
Partially fixes #381175
2006-12-01 10:27:54 +00:00