Commit graph

6819 commits

Author SHA1 Message Date
Benjamin M. Schwartz
d8a33f094c theoraenc: Add property for speed level control
Add property "speed-level" to control the amount of motion searching
the encoder does.  This is only available in libtheora >= 1.0 and
will silently fail with earlier libraries.  Fixes: #572275.

Signed-off-by: David Schleef <ds@schleef.org>
2009-02-19 12:38:57 -08:00
Edward Hervey
a490b3f2dd video: Fix 'Since' tags 2009-02-19 17:40:45 +01:00
Edward Hervey
c44b067817 video: Add flags for interlaced video along with convenience methods for interlaced caps.
These three flags allow all know combinations of interlaced formats. They should
only be used when the caps contain 'interlaced=True'.

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

Rename GstRTSPChannel to GstRTSPWatch to avoid confusing with the RTSP channels.
2009-02-19 15:55:07 +01:00
Edward Hervey
02f9079d6b Add some more mappings for h264 in riff 2009-02-19 13:24:39 +01:00
Wim Taymans
66995db940 Add new RTSP symbols to def files
Add the new RTSP symbols to the windows def file.
2009-02-19 10:49:56 +01:00
Wim Taymans
e5d8551552 Add method to install callbacks on appsink
Based on pacth by Martin Samuelsson <martin dot samuelsson at axis dot com>
Fixes #571299.

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

Add a unit test for appsink.

Clean up some of the appsink docs.

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

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

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

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

Fixes #553295, crash with fuzzed AVI file.
2009-02-11 16:58:18 +00:00
Tim-Philipp Müller
2a89ee9dd3 Make git ignore backup files. 2009-02-11 16:58:17 +00:00
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
Christian Schaller
67948a4027 Update spec file with latest additions and changes, most noteably the move of appsrc appsink into -base 2009-02-10 18:29:22 +00:00
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
Peter Kjellerstedt
430eea3016 gstrtspmessage: Minor documentation correction.
Corrected documentation about what needs to be freed after calling
gst_rtsp_message_new(), gst_rtsp_message_new_request(),
gst_rtsp_message_new_response() and gst_rtsp_message_new_data().
2009-02-10 17:37:06 +01:00
Antoine Tremblay
fc23037a9a alsamixer: Fix race condition that made alsamixer not working properly
This is due to race conditions between functions that
modified the mixer like set_volume and
snd_mixer_handle_events since the handle_events
can now be called at any time.

Fixed by adding locking around any snd_mixer call
since even read functions can modify the mixer stucture, since
alsa likes to clear it's values before reading new ones.

The favorite race condition seemed to be that set_volume
called read_elem (in alsalib) that reset the volumes to
0 and then read them with read_x_volume. This read looped
on each channel and as the race condition occured the
channels value could be anything , most of the time
it was 0. Thus no value was read or only the value of
one channel was and the volume was reset to 0.

Fixes bug #478512.
2009-02-10 11:00:12 +01:00
Edward Hervey
6e7fe156d8 Bump revision to use for common submodule. 2009-02-09 12:02:21 +01:00
Stefan Kost
26e4e3dcfe xvimagesink: do not call _xwindow_clear on ready->paused.
Calling clear at that transition does things like stopping xvideo (which is not
running at that time) and also clearing anything what the application might have drawn.
This breaks handle-expose and autopaint-colorkey features.
2009-02-05 15:51:25 +02:00
Wim Taymans
76112f9f04 RTSPRange: Add method to serialize ranges
Add gst_rtsp_range_to_string() to serialize a GstRTSPRange to a string that can
be used by a server.
API: GstRTSPRange::gst_rtsp_range_to_string()
2009-02-04 17:03:52 +01:00
Wim Taymans
4bb5722f1a GstRTSPUrl: Add some const to methods
Add const to the methods that do not modify the object.
2009-02-04 13:16:48 +01:00
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
Wim Taymans
ad1dea3122 Add more g_return_if_fail() calls
Check that we have a valid file descriptor before entering certain functions in
order to avoid undesirable situations.
Add some more debugging in the connect method.
2009-02-04 11:18:31 +01:00
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
Wim Taymans
314d852e48 Add new methods
Add new methods to the windows def file.
2009-02-03 10:52:15 +01:00
Tim-Philipp Müller
95d6fb0501 pbutils: remove duplicate detail strings when calling the external codec installer
It doesn't make sense to ask installers for the same codec or element twice, so filter out duplicate requests before calling the external helper script and make the unit test check this works right. Fixes #567636.
2009-02-02 17:34:23 +00:00
Stefan Kost
486fe43cb9 Add a FIXME 0.11. Make the log message a bit more detailed and add comments. 2009-02-02 18:05:42 +02:00
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
e7f910e0fc Fix buffer_alloc in ximagesink
Remove some useless debug info that reported wrong image sizes.

When upstream does not accept out suggested size, fall back to allocating an
image of the requested width/height instead of the currently configured size.
The problem is that an image is reused from the pool because the width/height
match but the caps on the new buffer are the requested caps with possibly
different height/width resulting in errors.
2009-02-02 13:30:42 +01:00
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
Wim Taymans
35cec4c006 Fix string leak in rtspmessage
when we remove a header field from a message we must free the value associated
with the key to avoid a memory leak.
2009-02-02 10:09:07 +01:00
Stefan Kost
a6bc6220d0 Its "Base Library" and not just "Library". 2009-01-31 18:45:47 +02:00
Stefan Kost
950d0c0a7d Link to the class, as we can't link to the members yet. 2009-01-31 18:44:32 +02:00
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