Commit graph

6884 commits

Author SHA1 Message Date
Sebastian Dröge
a64caea0bd videofilter: Add a default get_unit_size function
This returns the correct values for all formats that are handled by
GstVideoFormat and makes all the custom get_unit_size functions in
many elements unnecessary.
2009-06-16 19:38:17 +02:00
Wim Taymans
33837d420c rtsp: add Timestamp header field
fixes #585994
2009-06-16 18:57:20 +02:00
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
Tim-Philipp Müller
70089160f8 audiosink, audiosrc: do the class_ref()s in the right class_init functions
Spotted by Philip Jägenstedt. Hopefully fixes #585970 for real.
2009-06-16 14:14:26 +01:00
Tim-Philipp Müller
3767cb6005 audiosink,audiosrc: ref the audio ring buffer class and type in class_init
Hack around thread-safety issues in GObject and our racy _get_type()
functions (we could easily fix the _get_type() functions, but we still
need to hack around the GObject class races until we require a newer
GLib version, I think).
2009-06-15 15:39:09 +01:00
Wim Taymans
a5491ba218 audiosrc: return FALSE when receiving a SEEK event
When receiving a seek event, return FALSE as we don't implement seeking.
2009-06-15 12:57:39 +02:00
Sebastian Dröge
79adfa544d Don't use deprecated GTK API
Fixes bug #585758.
2009-06-15 11:07:10 +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
Peter Kjellerstedt
73dd8236ce rtsp: Use a more consistent naming of GstRTSPRec variables. 2009-06-15 09:28:34 +02:00
Peter Kjellerstedt
ff38999c8b rtsp: Call message_sent() callback for all sent messages.
Previously the messages_sent() callback was only called for messages
which had a CSeq, which excluded all data messages. Instead of using the
CSeq as ID, use a simple index counter.
2009-06-15 09:28:13 +02:00
Tim-Philipp Müller
12134979a2 oggdemux: post/send tags with the container-format tag
For this to work properly, theoradec and vorbisdec need to put
tag events received from upstream into the pending_events list
so they get pushed out after any newsegment event, not before.
2009-06-14 22:13:41 +01:00
Sebastian Dröge
81a0a98611 Don't use deprecated GTK API
Fixes bug #585758.
2009-06-14 20:32:03 +02:00
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
22cdc527a5 seek: add shuttle controls 2009-06-12 13:55:33 +02:00
Wim Taymans
8e71d0587b example: fix compile 2009-06-12 13:55:02 +02:00
Wim Taymans
54dc7b963f examples: build the stepping2 example 2009-06-12 13:52:25 +02:00
Wim Taymans
6a7d0ebf2a playsink: update for new step API 2009-06-12 13:52:02 +02:00
Wim Taymans
acdb88ec6f oggdemux: do reverse seeks more accurate
For reverse seeking with the accurate flag set, try to be more precise by
seeking a little bit after the requested position.
2009-06-12 13:44:26 +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
Wim Taymans
a9c82f9472 ringbuffer: handle border cases in resampler 2009-06-11 19:13:28 +02:00
Jan Schmidt
b2930f24b0 docs: Update common. Use upload-doc.mak instead of upload.mak 2009-06-11 14:14:12 +01:00
Wim Taymans
8bbf2e8a32 docs: fix typo 2009-06-11 12:39:19 +02:00
Wim Taymans
69b7fb3845 baseaudiosink: reset accum when dropping samples
When we are resampling and we drop samples because we paused, reset the accum
counter because it's now invalid.
2009-06-11 12:38:35 +02:00
Jan Schmidt
c1bc55a4f5 docs: Fix a couple of warnings from the docs build. 2009-06-11 11:16:15 +01:00
Tim-Philipp Müller
249d9b4aa1 Don't include config.h multiple times when build audio testchannel app.
Fixes build problem on win32 (#585075).
2009-06-10 21:37:29 +01:00
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
e01fab3ace rtsp: add some more docs 2009-06-09 22:00:53 +02:00
Peter Kjellerstedt
263c5b227b rtsp: Avoid a compiler warning. 2009-06-09 18:24:55 +02:00
Peter Kjellerstedt
dfc57e3f8a rtsp: Updated documentation for GstRTSPResult.
Moved GST_RTSP_ELAST to be last in the documentation to match the actual
enum values.
2009-06-09 18:23:28 +02:00
Tim-Philipp Müller
07055a9297 autogen: remove -Wno-portability from here
as it is in configure.ac now.
2009-06-09 15:34:00 +01:00
Peter Kjellerstedt
9c40eeeb4c rtsp: Plug a memory leak.
Free memory related to any partially read and/or written RTSP messages.
2009-06-09 16:28:20 +02:00
Wim Taymans
38e59ec75d baseaudiosink: no need to cause discont when clipping
Remove the discont-when-clipping hack now that basesink provides us with
correctly clipped samples when stepping.
2009-06-09 12:09:15 +02:00
Wim Taymans
cb4952fc2e audiosink: don't align when we clip
Don't align samples when they were clipped. Not entirely correct but better than
nothing for now.
2009-06-08 17:26:59 +02:00
Wim Taymans
d3d661ec7f examples: add stepping example in PLAYING
Add stepping example in PLAYING, audio is a bit distorted because basesink does
not provide good clipping info yet.
2009-06-08 16:41:58 +02:00
Edward Hervey
ee3b251234 pbutils: Add description for hdv/aux-* formats. 2009-06-08 10:25:00 +02:00
LRN
30103e736d Added libgstbase to schro's LIBADD
Fixes #585079
2009-06-07 22:01:50 +02:00
Tim-Philipp Müller
5da78c8489 libgsttag: don't extract genres from empty ID3v1 tags
If we don't have any other info, don't try to interpret the
genre field. In particular we don't want to interpret a genre
of 0 as 'Blues' if no other fields are set and the entire tag
is just empty.
2009-06-06 12:04:12 +01:00
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
Peter Kjellerstedt
2dbd8702dd rtsp: Fixed a typo. 2009-06-05 14:06:17 +02:00
Peter Kjellerstedt
de18ad458f rtsp: Remove an unused variable. 2009-06-05 14:05:54 +02:00
Peter Kjellerstedt
b0a9848524 rtsp: Removed duplicate initialization of conn->writefd. 2009-06-05 13:59:14 +02:00
Peter Kjellerstedt
0167e3589d rtsp: Use #defined status codes. 2009-06-05 13:55:08 +02:00
Peter Kjellerstedt
c1a6644a18 rtsp: Correct gen_tunnel_reply().
Prevent gen_tunnel_reply() from generating an incomplete response
in case an error response code is given.
2009-06-05 13:53:29 +02:00
Tim-Philipp Müller
b2783a5c5a configure: remove AC_C_INLINE which is not needed and causes problems with MSVC
See #584835. Also update win32 files while we're at it.
2009-06-05 10:57:44 +01:00
Sebastian Dröge
28b366db00 playbin2: API: Add {audio,video,text}-tags-changed signals
Fixes bug #584686.
2009-06-04 08:59:28 +02:00
Tim-Philipp Müller
20fb58be19 vorbisdec: don't put invalid bitrate values into the taglist
Bitrates are stored as 32-bit signed integers in the vorbis
identification headers, but seem to be read incorrectly,
namely as unsigned 32-bit integers, into the vorbis structure
members which are of type long, which makes our check for
values <= 0 fail with files that put -1 in there for unset
values.
2009-06-03 20:42:39 +01:00