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Author SHA1 Message Date
Michael Smith
92560517e8 Merge branch 'master' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-base 2012-10-03 10:45:26 -07:00
Michael Smith
a29c4f9489 meta registration: use g_once functions to register these threadsafely. 2012-10-03 10:44:59 -07:00
Tim-Philipp Müller
81097f485a playback: class_ref() some types so we can create multiple playback elements at the same time 
Should fix "cannot register existing type `GstPlaybinSelectorPad'" warnings
and subsequent errors when creating multiple players at the same time.

Conflicts:
	gst/playback/gststreamselector.c
 2012-10-03 11:48:25 +01:00
Sebastian Dröge
9d59b789c7 videodecoder: Fix unused variable compiler warning if debugging is disabled 2012-10-02 09:29:49 +02:00
Sebastian Pölsterl
e8fed7f04b rtsp: mark url argument of gst_rtsp_url_parse() as out arg 
https://bugzilla.gnome.org/show_bug.cgi?id=685242
 2012-10-01 22:36:06 +01:00
Olivier Crête
531a5af30c videodecoder: Also use the object lock to protect the output_state 
Hold both the stream and the object lock to modify the output_state,
this way it can be safely modified while hold either one or the other.

Also, only hold the object lock in the query

https://bugzilla.gnome.org/show_bug.cgi?id=684832
 2012-10-01 14:43:29 -04:00
Wim Taymans
370d336e9e docs: update for 1.0 2012-10-01 11:58:36 +02:00
Alban Browaeys
579458f613 encodebin: muxer sink pad is not always a request pad 
GstId3Mux sink pad is an always (static) pad. Thus releasing it
as if a request pad triggers:
(sound-juicer:11826): GStreamer-CRITICAL **:
gst_element_release_request_pad: assertion `GST_PAD_PAD_TEMPLATE (pad)
== NULL || GST_PAD_TEMPLATE_PRESENCE (GST_PAD_PAD_TEMPLATE (pad)) ==
GST_PAD_REQUEST' failed

https://bugzilla.gnome.org/show_bug.cgi?id=685110
 2012-09-30 15:08:17 +01:00
Tim-Philipp Müller
80e45be3d0 appsrc: fix max-latency property getter 
Was returning the min-latency value.
 2012-09-29 21:42:46 +01:00
Tim-Philipp Müller
6842698f0d Purge all references to liboil 
And remove unused ffmpegcolorspace tests in the process.

https://bugzilla.gnome.org/show_bug.cgi?id=673285
 2012-09-29 11:47:52 +01:00
Mark Nauwelaerts
4adfff03ef video{de,en}coder: fix missing timestamp estimating 
... by having some more timestamp tracking in a private frame field.
Not doing so would lead to (a.o.) losing the needed minimum timestamp in
an earlier sent frame.
 2012-09-28 13:59:24 +02:00
Sebastian Dröge
c4fb8d1e69 basetextoverlay: Correctly handle empty text buffers 2012-09-27 12:41:28 +02:00
Mark Nauwelaerts
dc2f2c9a40 videodecoder: use oldest frame DTS to estimate missing outgoing PTS 2012-09-27 11:31:34 +02:00
Mark Nauwelaerts
dbc89e3ab6 videoencoder: use oldest frame PTS to estimate missing outgoing DTS 2012-09-26 16:32:37 +02:00
Mark Nauwelaerts
d247301aec videoencoder: incoming buffer DTS is irrelevant 
... and bogus anyway if PTS != DTS
 2012-09-26 16:32:37 +02:00
Wim Taymans
65b06e18ac test: fix for new-sample signature 
The new-sample signal expects a GstFlowReturn as a result.
Add support for external subtitles as well.
 2012-09-26 13:31:50 +02:00
Mark Nauwelaerts
6973a66813 videoencoder: clip input buffers to current input segment 
... rather than to output segment, which will only be set
to current input segment if some output is produced
(coming from non-clipped input).

Also fixup debug message.
 2012-09-25 17:19:15 +02:00
Sebastian Dröge
a3878f8bb8 videoconvert: Set correct plugin metadata 2012-09-25 13:16:45 +02:00
Tim-Philipp Müller
b0baf45355 Back to development (bug fixing) 2012-09-24 16:46:44 +01:00
Tim-Philipp Müller
146ca8e3bf Release 1.0.0 2012-09-24 13:38:11 +01:00
Tim-Philipp Müller
62c111f1e4 videodecoder: don't take STREAM_LOCK on upstream events 
Don't try to take STREAM_LOCK on upstream events such as QOS.
Protect qos-related variables with object lock instead. Fixes
possible deadlock when shutting down in certain situations.

https://bugzilla.gnome.org/show_bug.cgi?id=684658
 2012-09-24 10:56:35 +01:00
Thiago Santos
386206e627 videotestsrc: keep track of the correct running time after renegotiations 
Need to store the old running time and frame numbers when renegotiating and
start from 0 again when a new caps is set, preventing that framerate changes
cause timestamping issues.

For example, if a stream pushed 10 buffers on framerate=2/1, its
running time will be 5s. If a new framerate of 1/1 is set, it would
make the running time go to 10s as it would count those 10 buffers
as being sent on this new framerate.

Fixes camerbin unit test.

https://bugzilla.gnome.org/show_bug.cgi?id=682973
 2012-09-23 17:48:56 +01:00
Tim-Philipp Müller
cec6d634b6 adder: send stream-start event, and send caps event after stream-start 
Delay sending of caps event so that it is sent only after
the stream-start event.
 2012-09-23 13:31:17 +01:00
Tim-Philipp Müller
1c22b1fe11 oggmux: send stream-start event 2012-09-23 13:27:27 +01:00
Tim-Philipp Müller
21ca8cc771 Automatic update of common submodule 
From 4f962f7 to 6c0b52c
 2012-09-22 16:07:35 +01:00
Tim-Philipp Müller
e072bd6130 oggmux: fix up previous commit 
Was missing the header file change.
 2012-09-21 16:10:27 +01:00
Tim-Philipp Müller
5890a4a803 oggmux: send a segment event at the beginning 2012-09-21 15:58:07 +01:00
Sebastian Dröge
1e8f5a0b06 videodecoder: Update comments about forwarding/not-forwarding serialized events immediately 2012-09-20 10:04:30 +02:00
Olivier Crête
ebae8ffa71 videodecoder: Protect all accesses to priv->output_frame with the stream lock 
Fixes segfault as queries/events can happen after a reset
 2012-09-19 21:16:01 -04:00
Andreas Frisch
6dd8302029 tests: port playbin-text example to 1.0 api 
https://bugzilla.gnome.org/show_bug.cgi?id=684084
 2012-09-19 16:41:48 +01:00
Arun Raghavan
9f9718715a audio: Explicitly specify endianness for IEC 61937 payloading 
This is required since some systems (DirectSound and OS X) manage the
final byte order themselves.

https://bugzilla.gnome.org/show_bug.cgi?id=678021
 2012-09-19 09:15:16 +05:30
Mark Nauwelaerts
17e3dc3357 audioresample: mark semi-unused variable 
../../../gst-plugins-base/gst/audioresample/gstaudioresample.c: In function 'gst_audio_resample_dump_drain':
../../../gst-plugins-base/gst/audioresample/gstaudioresample.c:729:9: warning: variable 'in_len' set but not used [-Wunused-but-set-variable]
 2012-09-18 13:16:39 +02:00
Tim-Philipp Müller
993014c8f5 Release 0.11.99 2012-09-17 17:57:19 +01:00
Tim-Philipp Müller
5a5344633c docs: update 2012-09-17 17:57:09 +01:00
Tim-Philipp Müller
06777095e8 examples: make snapshot example actually compile and work 
https://bugzilla.gnome.org/show_bug.cgi?id=684063
 2012-09-17 16:19:52 +01:00
Tim-Philipp Müller
5e0dfec62c Remove -DGST_USE_UNSTABLE_API 2012-09-17 16:05:37 +01:00
Javier Jardón
f0d3f33540 tests/examples/snapshot/snapshot.c: get caps from the sample 
pull-preroll signal returns a GstSample, not a GstBuffer

https://bugzilla.gnome.org/show_bug.cgi?id=684063
 2012-09-17 16:05:37 +01:00
Sebastian Dröge
b19944d1e4 gst: Update for link/unlink function API change 2012-09-17 13:24:52 +02:00
Tim-Philipp Müller
1795039dad docs: update docs and fix build a bit more 
Don't try to include plugin that doesn't exist any longer
(merged into the playback plugin).
 2012-09-17 12:07:30 +01:00
Christian Fredrik Kalager Schaller
a6d6b1954d Update spec file with latest changes and switch to F18 package naming 2012-09-15 22:08:30 +02:00
Mark Nauwelaerts
e491d24341 use gst_element_factory_get_metadata to replace obsolete API 2012-09-15 18:57:09 +02:00
Mark Nauwelaerts
c629a44162 replace gst_tag_list_free with gst_tag_list_unref 2012-09-14 17:53:21 +02:00
Mark Nauwelaerts
f7c247b6a3 replace gst_element_class_set_details_simple with gst_element_class_set_metadata 2012-09-14 17:02:59 +02:00
Wim Taymans
a57198a0ba audio: improve property description 
Improve the description of the latency-time and buffer-time properties in the
audio sink and source.
 2012-09-14 16:08:50 +02:00
Sebastian Dröge
6e33f2d464 audiodecoder: Don't output an (unreffed) buffer in error cases 2012-09-14 14:54:22 +02:00
Wim Taymans
24bab1e5a8 fix for appsink GstFlowReturn 2012-09-14 13:39:20 +02:00
Wim Taymans
e46b45b0b8 appsink: add GstFlowReturn from signal handler 
Expect a GstFlowReturn from the signal handler, just like from the callback.
Also use the return value.
 2012-09-14 13:31:36 +02:00
Wim Taymans
acb3aeebd4 fix caps 2012-09-14 13:22:31 +02:00
Andreas Frisch
6e469b2ac5 playbin: subtitleoverlay: don't segfault in incorrectly init'ed plugins 
https://bugzilla.gnome.org/show_bug.cgi?id=683865
 2012-09-14 08:49:47 +01:00
Tim-Philipp Müller
77c3a225c8 Back to development 2012-09-14 02:57:01 +01:00