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6711 commits 44 branches 1510 tags 328 MiB
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1388 commits

Author SHA1 Message Date
Michael Smith
8f6399f109 riff: support UYVY raw 4:2:2 in riff. 2009-05-11 14:04:16 -07:00
Andy Wingo
9f74ce745f Revert "add can-activate-pull property to baseaudiosink" 
This reverts commit c4074a2ee4.
 2009-04-29 11:18:42 +02:00
Andy Wingo
219a31fa3c Revert "[baseaudiosink] add docs for can-activate-pull" 
This reverts commit 416ce16f26.
 2009-04-29 11:18:33 +02:00
Andy Wingo
416ce16f26 [baseaudiosink] add docs for can-activate-pull 
* gst-libs/gst/audio/gstbaseaudiosink.c: Add documentation for
  can-activate-pull.
 2009-04-28 18:48:33 +02:00
Andy Wingo
c4074a2ee4 add can-activate-pull property to baseaudiosink 
* gst-libs/gst/audio/gstbaseaudiosink.c: Add can-activate-pull property
  to baseaudiosink.
 2009-04-28 18:28:50 +02:00
Tim-Philipp Müller
8efe6108c4 cddabasesrc: fix posting of discid tags after MERGE_MODE_REPLACE_ALL changes in core 
Don't use REPLACE_ALL merge mode when that's not really what we want,
as now that REPLACE_ALL actually does what it's supposed to do in
core, we drop tags we wanted to keep, such as the various disc id
tags. Add unit test for this as well. Fixes #579463.
 2009-04-19 18:15:28 +01:00
Tim-Philipp Müller
418760cf74 rtspconnection: don't use GLib-2.16 API, we require only 2.14 
Fixes #579267.
 2009-04-17 10:35:34 +01:00
Wim Taymans
32904de58f baseaudiosink: don't unparent the ringbuffer 
when going to NULL, don't unparent the ringbuffer because we don't support going
back to 0 very well yet.
Fixes #579203
 2009-04-17 11:03:32 +02:00
Olivier Crete
d927114ef8 RTCP: don't fail when retrieving invalid PT 
We can't meaningfully assert on valid packet types so just return the type as it
is. Update the comments to reflect this.

Fixes #579192.
 2009-04-17 10:53:10 +02:00
Wim Taymans
f83f57b648 app: add trivial cast macros 
Add trivial cast macros for appsrc and appsink. Mark them as being since 0.10.23
and add the macros to the standard macros in the docs.

Fixes #579130
 2009-04-16 12:14:43 +02:00
Sebastian Dröge
a6cf0c8f06 video: Fix typo in the docs 2009-04-15 15:35:59 +02:00
Sebastian Dröge
a1d8cfde9d video: Add support for YVYU YUV colorspace 2009-04-15 14:53:47 +02:00
Tim-Philipp Müller
75acca2835 docs: fix hyperlink and move fft attribution to the right place 2009-04-15 00:19:19 +01:00
Stefan Kost
ab24d9d65c log: use G_GUINT64_FORMAT instead of llu 2009-04-15 00:02:39 +03:00
Josep Torra
71ab187355 RTSP: add missing headers for WMS RTSP 
Add missing headers related to Windows Media RTSP extension.
Fixes #578942
 2009-04-14 18:31:52 +02:00
Tim-Philipp Müller
9f23b82b2c Give credit to Mark Borgerding (kissfft author) 
and add myself to AUTHORS as well. Fixes #575638.
 2009-04-14 17:11:19 +01:00
Johann Prieur
86edcadc43 RTCP: add beginnings of Feedback messages 
Add the beginnings of parsing and constructing Feedback messages.
Fixes #577610.
 2009-04-14 16:45:20 +02:00
Wim Taymans
dffd1bcc97 baseaudiosrc: adjust the internal timestamp 
Adjust the internal timestamp before comparing it against the adjusted clock
time.
Fixes #578506
 2009-04-14 13:16:14 +02:00
Wim Taymans
0c4c1410f9 baseaudiosink: use new clock time methods 
Use the unadjusted internal clock times to calculate the internal/external
offset when calibrating the clock.

When going to NULL, unparent and free the ringbuffer, like we do in the source
element.
See #578506
 2009-04-14 13:12:59 +02:00
Wim Taymans
4231d54823 audioclock: add methods for the internal offset 
Add two methods for getting the unadjusted time of the clock and one for
adjusting an internal time. We will need these methods for correctly handling
the time after a gst_audio_clock_reset().

Add a debug category and some debug lines to the audio clock.

API: gst_audio_clock_get_time()
API: gst_audio_clock_adjust()
API: GST_AUDIO_CLOCK_CAST()
 2009-04-14 13:08:52 +02:00
Wim Taymans
251f152c20 baseaudiosink: use the internal clock time 
We can't assume that the internal clock time is the same as the function we
installed on our provided clock because somebody might have changed it.
 2009-04-10 21:50:55 +02:00
Martin Samuelsson
ee03bf5379 appsink: make callbacks return GstFlowReturn 
Make the new_buffer and new_preroll callbacks return a GstFlowReturn so that
errors can be reported properly.
Fixes #577827.
 2009-04-09 23:46:17 +02:00
Wim Taymans
e6798c5cce ringbuffer: allow for custom commit functions 
Allow subclasses to override the commit method.
 2009-04-09 18:04:44 +02:00
Wim Taymans
cae2981f83 baseaudiosink: fix a small glitch after pause 
After we pause the stream and interrupt the writeout to the ringbuffer, also adjust
the amount of output samples we consumed. We can't do this reliably with the
current API when we are doing trick modes but we can do the right thing for
normal playback.
 2009-04-08 18:06:54 +02:00
Stefan Kost
ff9ee1dc5a audiofilter: don't leak pad-template 
gst_element_class_add_pad_template() does not take ownership.
 2009-04-07 22:39:07 +03:00
Edward Hervey
2555eeb737 navigation/v4l: Don't use g_return_val_if_fail for computed/used values. 2009-04-04 16:28:14 +02:00
Wim Taymans
88110ea67e rtsp: use fully qualified urls when using a proxy 
Use a fully qualified url when specifying the url for tunneled requests through
a proxy.
See #573173
 2009-04-02 22:28:55 +02:00
Jan Schmidt
033e654172 navigation: Extend the navigation interface 
Add support for a set of standard commands that can be queried and executed to
support applications like DVD. Add query construction and parsing functions.
Add new messages that can be sent on the bus to provide notifications related
to commands, multiangle changes, and button highlight activity.
Add some helper functions to parse the existing GstNavigation events that
elements might receive.
Document it all and add unit tests.
 2009-04-02 12:21:18 +01:00
Wim Taymans
eed784b372 rtsp: fix little typo in the comments 2009-04-01 09:03:35 +02:00
Tim-Philipp Müller
fc8c5cba15 rtspconnection: make gst_rtsp_watch_queue_message() thread-safe 
People might queue messages from a thread other than the thread in which
the main context which this watch is attached is iterated from, so use
a GAsyncQueue instead of a GList, so g_list_append() doesn't trample
over list nodes just freed in the other thread. This just fixes issues
I've had with gst-rtsp-server. We might need more locking in various
places here.
 2009-03-31 18:30:57 +01:00
Tim-Philipp Müller
dfe96ce618 rtsp: clear the entire builder structure 
And use structure instead of variable with sizeof when
clearing the rtsp message structure, for clarity.
 2009-03-31 18:30:48 +01:00
Tim-Philipp Müller
dd9f077177 docs: fix typo in gst_rtsp_message_unset() API docs 2009-03-31 18:30:48 +01:00
Wim Taymans
8b37dc3eb8 rtsp: add support for proxies 
Add suport for proxy servers. Currently only used for tunneled HTTP
connections without authentication.
 2009-03-31 19:00:00 +02:00
Wim Taymans
8be68f983c Revert "rtsp: reset whole message (was sizeof pointer instead of sizeof type)" 
This reverts commit 79de0b8d67.
 2009-03-31 18:57:08 +02:00
Stefan Kost
79de0b8d67 rtsp: reset whole message (was sizeof pointer instead of sizeof type) 2009-03-31 12:27:09 +03:00
Jan Schmidt
43788e4796 doc: Fix a typo in the GstMixer docs 2009-03-31 00:58:24 +01:00
Wim Taymans
0d3d3026d2 rtsp: start CSeq counting from 1 instead of 0 
Start counting from 1 instead of 0 as this is what most other clients
seem to do.
 2009-03-25 16:37:28 +01:00
Wim Taymans
1081ae59eb rtsp: add ETag and If-Match headers 
Add new headers, we need them for RealMedia support.
 2009-03-25 16:36:14 +01:00
Tim-Philipp Müller
0267e79778 audiosrc: improve 'Dropped n samples' warning message 2009-03-25 11:27:44 +00:00
Sebastian Dröge
108ead73c8 rtsp: Use GLib base64 functions and deprecate gst_rtsp_base64_encode 
This also fixes another instance of CVE-2008-4316.
 2009-03-17 22:53:44 +01:00
Wim Taymans
f4b7cbbf16 rtsp: fix resolving of hostnames 
We were returning a pointer to a stack variable with the resolved hostname,
which doesn't work.
return a copy of the resolved ip address instead.
Fixes #575256.
 2009-03-13 16:19:41 +01:00
Wim Taymans
91b2d71da0 appsrc: release lock in _eos flushing case 
Release the mutex when we are flushing in gst_app_src_end_of_stream()
Fixes #574964.
 2009-03-13 15:16:44 +01:00
Jan Schmidt
566583e871 vorbistag: Protect memory allocation calculation from overflow. 
Patch by: Tomas Hoger <thoger@redhat.com> Fixes CVE-2009-0586
 2009-03-12 15:02:07 +00:00
Wim Taymans
0e2157029e rtsp: fix parsing of the timeout parameter 
--
 2009-03-11 18:45:59 +01:00
Wim Taymans
b674584e97 rtsp: fix g_return condition 
when parsing a data message, we require a data message.
 2009-03-11 17:29:41 +01:00
Wim Taymans
18f612ffa9 rtsp: free the right string. 
Free the key value before we remove the header item from the array. The item we
retrieved from the array is only valid until we remove it from the array.
 2009-03-11 14:09:54 +01:00
Wim Taymans
16225d45be rtsp: keep track of amount of decoded bytes 
Keep track of the actual amount of decoded bytes, which can be less than 3 when
we decode the last bits of a base64 message.
 2009-03-11 14:09:54 +01:00
Wim Taymans
f964c0fc38 rtsp: only add ports when not using TCP 
Only add the port numbers in the transport string when we are using udp or
multicast.
 2009-03-09 13:53:41 +01:00
Wim Taymans
bc54a5f9a0 rtsp: use gstreamer dump mem 
--
 2009-03-09 13:53:15 +01:00
Wim Taymans
3a72044a22 rtsp: use glib base64 encoder 
--
 2009-03-09 13:51:48 +01:00