Commit graph

6692 commits

Author SHA1 Message Date
Sebastian Dröge
dc9a1b945b vorbis: Rename vorbis*.h to gstvorbis*.h to prevent name conflicts 2009-02-24 14:36:39 +01:00
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
30f0b8171f stress-playbin: make more flexible, e.g. also useful for playbin2 2009-02-24 13:30:07 +01:00
Wim Taymans
bb5e2d3f56 Match WSAStartup and WSACleanup correctly
Don't randomly call WSAStartup and WSACleanup but instead call the startup when
we create a connection and cleanup when we free it again. Because the internal
datastructure is refcounted, this should not cause any refcounting leaks when
the connection is managed correctly.
Fixes #562794.
2009-02-24 12:11:00 +01:00
Mark Nauwelaerts
bbd66c6baf playbin2/playsink: Set audiotee to PAUSED state in all cases. Fixes #565105. 2009-02-24 10:46:35 +01:00
David Flynn
00654ba62b Add srcdir to includes for out-of-source builds
When you use gstreamer uninstalled and build outside
the source tree, the includes need to be specified for
both the source tree and the build tree.

Signed-off-by: David Schleef <ds@schleef.org>
2009-02-23 11:01:11 -08:00
Jan Schmidt
8e08f6be68 Use shave for the build output 2009-02-23 13:19:50 +00:00
Edward Hervey
ab02950dcb win32: Add new symbol to libgstrtsp.def 2009-02-23 12:17:07 +01:00
Wim Taymans
6e560ae5d8 Add method for handling server requests
Add a receive_request so that extensions can react to server requests.
2009-02-23 10:57:08 +01:00
Sebastian Dröge
f14015567b Correctly cast to GstBuffer * before passing to gst_buffer_(copy|unref) 2009-02-22 19:20:40 +01:00
Sebastian Dröge
2ab2bbd82c theoraparse: Use the correct unref functions 2009-02-22 19:19:04 +01:00
Sebastian Dröge
8c74d858ba x(v)imagesink: Correctly cast to GstBuffer * before passing to gst_buffer_unref() 2009-02-22 19:18:41 +01:00
Sebastian Dröge
d659e8353d tagdemux: Unref the actual buffer instead of the memory address of the buffer 2009-02-22 19:12:00 +01:00
Jan Schmidt
9d91e56674 Automatic update of common submodule
From 5d7c9cc to 9cf8c9b
2009-02-22 15:47:53 +00:00
Edward Hervey
bd7c9ecba3 win32/common: Update .def files for recent API addition 2009-02-22 14:49:29 +01:00
Edward Hervey
83fe624025 tests: Fix indentation 2009-02-22 13:43:35 +01:00
Edward Hervey
5ce5433152 libs/video: Fix gst_video_format_new_caps* functions.
Only add a 'interlaced=True' property to caps *IF* it is interlaced, else
don't add anything.
2009-02-22 13:42:33 +01:00
David Schleef
32b1b7904d Automatic update of common submodule
From 80c627d to 5d7c9cc
2009-02-21 11:13:36 -08:00
Wim Taymans
15cd839f81 Improve key/value parsing
Improve header field parsing by keeping a ref to the key/value instead of
copying it into a local variable.
2009-02-20 17:26:40 +01:00
Wim Taymans
bb4310203a Add trailing \0 to message length
We always put a trailing 0 at the end of the message body. Reflect this fact in
the length of the message.
2009-02-20 12:35:53 +01:00
Wim Taymans
0ffd5e703a Don't parse headers for data messages
Don't try to parse the headers on a data message because they don't have
headers.
2009-02-20 09:52:16 +01:00
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