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Author SHA1 Message Date
Matej Knopp
f89d7ee7eb Appsink fixes 2011-12-05 09:34:50 +01:00
Tim-Philipp Müller
5440ae3c18 Suppress deprecation warnings in selected files, for g_static_rec_mutex_* mostly 
GStaticRecMutex is part of our API/ABI, not much we can do here
in 0.10 for most of these.
 2011-12-04 20:50:25 +00:00
Tim-Philipp Müller
0d98aa25b8 Work around deprecated thread API in glib master 
Add private replacements for deprecated functions such as
g_mutex_new(), g_mutex_free(), g_cond_new() etc., mostly
to avoid the deprecation warnings. We'll change these
over to the new API once we depend on glib >= 2.32.

Replace g_thread_create() with g_thread_try_new().
 2011-12-04 17:16:30 +00:00
Tim-Philipp Müller
6098442bd0 xmpwriter: update for thread API deprecations in glib master 2011-12-04 15:23:21 +00:00
Wim Taymans
1225aa9a78 update for basesink event handler changes 2011-12-02 22:24:43 +01:00
Tim-Philipp Müller
177525f89f Merge remote-tracking branch 'origin/master' into 0.11 
Conflicts:
	gst-libs/gst/netbuffer/gstnetbuffer.c
	gst/ffmpegcolorspace/avcodec.h
	gst/ffmpegcolorspace/gstffmpegcodecmap.c
	gst/ffmpegcolorspace/imgconvert.c
	gst/ffmpegcolorspace/imgconvert_template.h
	gst/ffmpegcolorspace/mem.c
	gst/playback/README
	gst/playback/gstplaybasebin.c
	gst/playback/gstplaybasebin.h
	gst/playback/gstplaybin.c
	sys/v4l/v4lmjpegsrc_calls.c
	sys/v4l/videodev_mjpeg.h
	tests/check/elements/gnomevfssink.c
 2011-12-02 11:10:17 +00:00
Piotr Fusik
14644457b0 various: typo fixes 
Fix typos in code and docs. Fixes. #658984
 2011-12-02 12:03:27 +01:00
Tim-Philipp Müller
ec0d3566bf Merge remote-tracking branch 'origin/master' into 0.11 
Conflicts:
	ext/alsa/gstalsasrc.c
	ext/alsa/gstalsasrc.h
	gst/adder/gstadder.c
	gst/playback/gstplaybin2.c
	gst/playback/gstplaysinkconvertbin.c
	win32/common/libgstvideo.def
 2011-12-02 00:07:39 +00:00
Wim Taymans
3deaa582d9 tags: make the tag functions return GstSample 
gst_tag_image_data_to_image_buffer() ->
   gst_tag_image_data_to_image_sample() And make it return a GstSample.
Store the image-type into the extra sample info.
Remove a deprecated tag
 2011-12-01 18:51:51 +01:00
Wim Taymans
59113af604 Use the new GstSample for snapshots 
Make appsink return a GstSample. Remove the pull_buffer_list method because it
is not very useful anymore.
Pass GstSample to the conversion function.
Update playbin2 and examples
 2011-12-01 16:53:11 +01:00
Wim Taymans
66d7151787 update marshal list 2011-12-01 15:54:49 +01:00
Wim Taymans
92ac25bdb3 videometa: add copy functions 
Without copy functions, the metadata is lost when we make a buffer copy such as
when we make a buffer writable.
 2011-12-01 15:45:28 +01:00
Wim Taymans
e064f9dbf6 appsrc: fix negotiation 
Remove old useless caps code.
Make a negotiate function and use the configured caps as the caps on the appsrc
pad. If nothing was configured, fall back to the parent implementation.
 2011-12-01 15:38:10 +01:00
Edward Hervey
e44db979f9 audio: Add audio-marshal.list to dist-ed files 2011-11-30 11:33:41 +01:00
Wim Taymans
47cbb230e9 audio: move audio interfaces 
Move the audio related interfaces to the audio library.
 2011-11-30 07:57:02 +01:00
Wim Taymans
4fb0f98bb9 encoding-profile: small cleanup in docs 2011-11-30 07:23:07 +01:00
Edward Hervey
5bc6ffcd8b video: Don't forget to install moved header files 2011-11-29 19:49:50 +01:00
Wim Taymans
871b306fce video: move some interfaces 
Move some interfaces to the video library
 2011-11-29 19:10:01 +01:00
Alessandro Decina
848711706b libgstvideo: minor fixes to key unit events 
Make out args to gst_video_event_parse_{downstream|upstream}_force_key_unit
optional, update libgstvideo.def and fix docs a bit.

API: gst_video_event_new_upstream_force_key_unit
API: gst_video_event_new_downstream_force_key_unit
API: gst_video_event_is_force_key_unit
API: gst_video_event_parse_upstream_force_key_unit
API: gst_video_event_parse_downstream_force_key_unit

https://bugzilla.gnome.org/show_bug.cgi?id=607742
 2011-11-29 09:15:59 +01:00
Andoni Morales Alastruey
df44e771f1 libgstvideo: Add force key unit events 2011-11-29 08:58:28 +01:00
Tim-Philipp Müller
0d87fd7146 Merge remote-tracking branch 'origin/master' into 0.11 
Conflicts:
	gst-libs/gst/fft/gstffts16.h
 2011-11-28 21:25:11 +00:00
Tim-Philipp Müller
0c056a04fe Merge commit '4a58223e4c824fedc024af435337a769e8ce593e' into 0.11 2011-11-28 21:20:10 +00:00
Philippe Normand
0a841f6712 fft: Bracket public headers 
This is especially needed if the gstfftw library is used from C++
code.

Fixes #665074
 2011-11-28 20:28:19 +01:00
Wim Taymans
5b868bd424 Update for indexable change 2011-11-28 18:24:03 +01:00
Wim Taymans
468d1dde89 audio: update for clock provider API change 2011-11-28 17:51:41 +01:00
Vincent Penquerc'h
ea78b060a7 Revert "libgstvideo: add a new API to handle QoS events and dropping logic" 
This reverts commit eb03323fb6.

*grumble* I managed to merge something I did not mean to.
 2011-11-28 13:26:53 +00:00
Vincent Penquerc'h
96374054ac various: fix pad template leaks 
https://bugzilla.gnome.org/show_bug.cgi?id=662664
 2011-11-28 13:09:02 +00:00
Vincent Penquerc'h
eb03323fb6 libgstvideo: add a new API to handle QoS events and dropping logic 
https://bugzilla.gnome.org/show_bug.cgi?id=658241
 2011-11-28 12:34:43 +00:00
Mark Nauwelaerts
4a58223e4c audioencoder: elaborate some documentation 2011-11-28 11:37:33 +01:00
Mark Nauwelaerts
9f57d91137 audiodecoder: add some documentation 2011-11-28 11:37:27 +01:00
Mark Nauwelaerts
856a5dd581 audiodecoder: really discard NULL decoded frame altogether 
... including any timestamp, rather than having that one influence base_ts.
 2011-11-28 11:37:23 +01:00
Tim-Philipp Müller
32b14c6ed3 Merge remote-tracking branch 'origin/master' into 0.11 
Conflicts:
	ext/vorbis/gstvorbisenc.c
	gst/playback/gstdecodebin2.c
	gst/playback/gstplaysinkconvertbin.c
	gst/videorate/gstvideorate.c
 2011-11-26 12:12:59 +00:00
Tim-Philipp Müller
a0639dad38 audio: remove unstable API guards from the audio decoder and encoder base classes 2011-11-25 13:11:54 +00:00
Edward Hervey
d94535832b gst-libs: Add --warn-all to introspection scanner 
And let's get fixing those docs :)
 2011-11-25 10:31:38 +01:00
Vincent Penquerc'h
44ec58b41b libgstriff: add a couple tags that need skipping 
Found in a sample in the wild, appears to be ID3 tag.

https://bugzilla.gnome.org/show_bug.cgi?id=660249
 2011-11-24 17:04:19 +00:00
Matej Knopp
817f39608c Fix printf format compiler warnings for OSX / 64bit 
https://bugzilla.gnome.org/show_bug.cgi?id=662607
 2011-11-22 01:00:59 +00:00
Wim Taymans
8fc2a21775 update for activation changes 2011-11-21 13:35:34 +01:00
Wim Taymans
d0bd5f04c0 update for new scheduling query 2011-11-18 17:58:58 +01:00
Wim Taymans
1ad4d20607 add parent to activate functions 2011-11-18 13:56:04 +01:00
Wim Taymans
285702a1a6 fix for scheduling mode rename 2011-11-18 12:37:10 +01:00
Wim Taymans
7afdff3575 Merge branch 'master' into 0.11 
Conflicts:
	gst-libs/gst/audio/gstaudiodecoder.c
 2011-11-17 17:07:41 +01:00
Wim Taymans
3a53451501 tag: update for new typefind 2011-11-17 16:15:46 +01:00
Wim Taymans
e302833e65 add parent to pad functions 2011-11-17 12:48:25 +01:00
Mark Nauwelaerts
69c2c46472 audioencoder: invalidate format info when setup negotiation failed 
... which ensures nothing subsequently tries to slip past _chain
and into a possibly improperly setup subclass.
 2011-11-16 19:03:47 +01:00
Vincent Penquerc'h
f17f918b75 audiodecoder: accept dropped buffers before we know the format 
This allows flacdec to not emit audio for headers, while allowing
the base audio decoder to keep its timestamps in sync.
 2011-11-16 16:54:03 +00:00
Wim Taymans
2202511e77 add parent to query function 2011-11-16 17:25:17 +01:00
Wim Taymans
28157e6f21 _query_peer_*() -> _peer_query_*() 2011-11-15 18:04:17 +01:00
Wim Taymans
026ec68f75 _peer_get_caps() -> _peer_query_caps() 2011-11-15 18:04:17 +01:00
Wim Taymans
7402d3a3d2 update for _get_caps() -> _query_caps() 2011-11-15 18:04:17 +01:00
Wim Taymans
ab9ffa93f5 change getcaps to query 
Add sink and src event functions in rtpbasepayload
Add query vmethod to rtpbasepayload.
 2011-11-15 18:04:16 +01:00
Vincent Penquerc'h
3e095382a1 audiodecoder: accept dropped buffers before we know the format 
This allows flacdec to not emit audio for headers, while allowing
the base audio decoder to keep its timestamps in sync.
 2011-11-15 13:29:31 +00:00
Robert Swain
a23dff1fbb audio: Remove some unused variables 2011-11-14 12:49:50 +01:00
Olivier Crête
82827df405 rtcpbuffer: Add feedback message types from RFC 5104 
These are Codec Control messages (CCM)

https://bugzilla.gnome.org/show_bug.cgi?id=658419
 2011-11-14 12:24:16 +01:00
Mark Nauwelaerts
38615abdd8 audiodecoder: improve reverse playback 
... by doing some more (reverse) timestamp interpolating and
refactoring downstream pushing.

Fixes #661983.
 2011-11-14 12:00:06 +01:00
Tim-Philipp Müller
046dc1097c tag: convert GstTagDemux's sometimes source pad to an always source pad 
Originally decodebin couldn't deal with that in 0.10, but now simply
setting the caps when we know them should be enough. Pad activation
mode switching might need some more testing/tweaking with the new
arrangement.
 2011-11-14 10:07:06 +00:00
Wim Taymans
fc04bcecbe fix docs 2011-11-14 10:46:56 +01:00
Tim-Philipp Müller
c76e5804b3 Update for GstURIHandler get_protocols() changes 2011-11-13 23:44:23 +00:00
Tim-Philipp Müller
455f337e3d gio, appsrc, appsink, cdaudiosrc: update for GstURIHandler API changes 2011-11-13 18:22:06 +00:00
Tim-Philipp Müller
4b0dce5148 Merge remote-tracking branch 'origin/master' into 0.11 
Conflicts:
	gst-libs/gst/audio/Makefile.am
	gst-libs/gst/audio/audio.h
	tests/examples/seek/jsseek.c
	tests/examples/seek/seek.c
	tests/icles/test-colorkey.c
 2011-11-13 13:36:29 +00:00
Tim-Philipp Müller
cd21e69913 audio: add GST_AUDIO_INFO_IS_VALID macro and use in audio decoder base class 
API: GST_AUDIO_INFO_IS_VALID
 2011-11-13 13:18:16 +00:00
Tim-Philipp Müller
394b1f8c3c audio: fix order in LIBADD 
Local libs must come first.
 2011-11-12 12:13:05 +00:00
Tim-Philipp Müller
756c9e2948 audio: fix order in LIBADD 
Local libs must come first.
 2011-11-12 11:58:59 +00:00
Tim-Philipp Müller
dfc13ec632 cdda: rename GstCddaBaseSrc to GstAudioCdSrc and move to libgstaudio 
Another mini-lib down, to make space for new mini libs.

Remove bogus copyright line while at it.
 2011-11-12 11:58:58 +00:00
René Stadler
8023733da8 video: init chroma-size and colorimetry members even if missing from caps 
This makes a TRUE return from gst_video_info_from_caps fully consistent with
gst_video_info_init.
 2011-11-11 19:57:25 +01:00
Wim Taymans
bdf3845498 rtsp: cleanup headers 
Add padding, fix indentation, remove deprecated stuff
 2011-11-11 19:35:33 +01:00
Wim Taymans
107d5a3d05 rtp: fix headers 
indent, add padding, remove old abidata
 2011-11-11 19:21:09 +01:00
Wim Taymans
370dca92d5 remove padding from interfaces 2011-11-11 19:16:54 +01:00
Wim Taymans
43eafea6f6 fix docs 2011-11-11 19:16:12 +01:00
Wim Taymans
fa897def26 mixertrack: fix docs 2011-11-11 19:14:26 +01:00
Wim Taymans
c42e257751 audio: fix docs 2011-11-11 19:13:52 +01:00
Wim Taymans
bfd417644d pbutils: clean up headers 
Add padding
indent
 2011-11-11 19:01:56 +01:00
Wim Taymans
7fb914d5b6 interfaces: clean up 
Remove deprecated bits
Fix FIXMES
Indent
Add padding
 2011-11-11 18:49:09 +01:00
Wim Taymans
40be2eec9f fft: fix headers 
More fft structure into .c file
indent headers
 2011-11-11 18:23:22 +01:00
Wim Taymans
b645287775 audio: fix headers 
Add const to some methods.
Add padding.
Add GType for GstAudioInfo and GstAudioFormatInfo.
Add new/copy/free for GstAudioInfo.
 2011-11-11 17:53:03 +01:00
Wim Taymans
b12aabc9da app: fix headers 2011-11-11 17:52:36 +01:00
Wim Taymans
06a6ab3e32 video: add support for max-framerate 
Add support for max-framerate in the video helpers and update the video
caps document.
 2011-11-11 13:14:21 +01:00
Wim Taymans
b14e3b9adc remove bogus file 2011-11-11 12:35:50 +01:00
Wim Taymans
5f1312b5d8 rename files to match object names 2011-11-11 12:32:23 +01:00
Wim Taymans
ccf511a5d4 rename BaseRTP -> RTPBase 2011-11-11 12:24:08 +01:00
Wim Taymans
a3416bc11f rename baseaudio* -> audiobase* 2011-11-11 12:00:52 +01:00
Wim Taymans
ee7072fe7e rename GstBaseAudio* ->GstAudioBase* 2011-11-11 11:52:47 +01:00
Wim Taymans
3d0ac3ded2 rename files to match contained objects 2011-11-11 11:33:15 +01:00
Wim Taymans
6511f36fdb audio: GstRingBuffer -> GstAudioRingBuffer 2011-11-11 11:21:41 +01:00
Wim Taymans
b81af23992 audio: rename internal audio ringbuffer 2011-11-11 10:54:39 +01:00
Wim Taymans
ad8f694ec6 remove bogus files 
They got somehow commited in 7012e88090
 2011-11-11 10:39:52 +01:00
Wim Taymans
e338792ab0 update for adapter api changes 2011-11-10 18:32:39 +01:00
Wim Taymans
fe766cf9f4 videosink: reset padding 2011-11-10 17:52:36 +01:00
Wim Taymans
ace51b689f rtsp: remove deprecated base64 library 2011-11-10 17:39:10 +01:00
Wim Taymans
f8ef57ca48 Merge branch 'master' into 0.11 2011-11-10 17:26:12 +01:00
Wim Taymans
24347217a5 rtp: fix de/payloaders 
gst_basertppayload -> gst_base_rtp_payload
Add pts/dts support in the depayloader
Remove old timestamp code
Add a default getcaps function so subclasses can chain up to it instead of
relying on the return value of the getcaps function.
 2011-11-10 17:18:00 +01:00
Vincent Penquerc'h
0d47c615ad baseaudiosink: make unsigned properties unsigned, not signed 2011-11-10 15:55:31 +00:00
Wim Taymans
57eaf388e0 audio: fix base class vmethods 2011-11-10 16:24:12 +01:00
Wim Taymans
ea9bc40bf9 audiosrc: avoid deadlock 2011-11-10 16:05:19 +01:00
Wim Taymans
1f8fe283f6 audioclock: remove _full version 2011-11-10 13:51:23 +01:00
Wim Taymans
f80d73468e appsink: fix header 2011-11-10 13:51:23 +01:00
Edward Hervey
3fa654b41c pbutils: Fix introspection annotations 
Fixes #663689
 2011-11-10 12:47:51 +01:00
Wim Taymans
d77c8cafee Merge branch 'master' into 0.11 
Conflicts:
	common
	ext/pango/gsttextoverlay.c
	gst-libs/gst/video/video.c
 2011-11-09 12:11:59 +01:00
Wim Taymans
372b9329b9 remove query types 2011-11-09 11:47:54 +01:00
Wim Taymans
308f6301a8 update for pad probe api changes 2011-11-08 11:08:21 +01:00
Stefan Sauer
e9629e37b7 video: log important details and fix format strings 
If we complain about wrong parameters passed, also log the actual value.
 2011-11-08 09:32:00 +01:00
Tim-Philipp Müller
d7fc45f42e docs: fix up some Since: markers 2011-11-07 23:05:44 +00:00
Wim Taymans
616e9b706e fix for new pad probe types 
Restore the previous behaviour by only blocking downstream items and not
upstream events.
 2011-11-07 17:10:48 +01:00
Wim Taymans
7ac25e9b26 Merge branch 'master' into 0.11 
Conflicts:
	common
	configure.ac
	gst-libs/gst/audio/gstbaseaudiosink.c
	gst/playback/gstdecodebin2.c
	gst/playback/gstplaysinkaudioconvert.c
	gst/playback/gstplaysinkaudioconvert.h
	gst/playback/gstplaysinkvideoconvert.c
	gst/playback/gstplaysinkvideoconvert.h
 2011-11-07 12:23:15 +01:00
Felipe Contreras
3df415d4c7 baseaudiosink: make discont-wait configurable 
Now we can configure how much time to wait before deciding that a
discont has happened.

Also, adds getter and setter to allow derived implementations to set
this value upon construction.

Suggestions and several improvements by Havard Graff.

Signed-off-by: Felipe Contreras <felipe.contreras@gmail.com>
 2011-11-07 11:58:46 +01:00
Felipe Contreras
0a111bf26e baseaudiosink: delay the resyncing of timestamp vs ringbuffertime 
A common problem for audio-playback is that the timestamps might not
be completely linear. This is specially common when doing streaming over
a network, where you can have jittery and/or bursty packettransmission,
which again will often be reflected on the buffertimestamps.

Now, the current implementation have a threshold that says how far the
buffertimestamp is allowed o drift from the ideal aligned time in the
ringbuffer. This was an instant reaction, and ment that if one buffer
arrived with a timestamp that would breach the drift-tolerance, a resync
would take place, and the result would be an audible gap for the
listener.

The annoying thing would be that in the case of a "timestamp-outlier",
you would first resync one way, say +100ms, and then, if the next
timestamp was "back on track", you would end up resyncing the other way
(-100ms) So in fact, when you had only one buffer with slightly off
timestamping, you would end up with *two* audible gaps. This is the
problem this patch addresses.

The way to "fix" this problem with the previous implementation, would
have been to increase the "drift-tolerance" to a value that was greater
than the largest timestamp-outlier one would normally expect.  The big
problem with this approach, however, is that it will allow normal
operations with a huge offset timestamp vs running-time, which is
detrimental to lip-sync. If the drift-tolerance is set to 200ms, it
basically means that lip-sync can easily end up being off by that much.

This patch will basically start a timer when the first breach of
drift-tolerance is detected. If any following timestamp for the next n
nanoseconds gets "back on track" within the threshold, it has basically
eliminated the effect of an outlier, and the timer is stopped.  If,
however, all timestamps within this time-limit are breaching the
threshold, we are probably facing a more permanent offset in the
timestamps, and a resync is allowed to happen.

So basically this patch offers something as rare as both higher
accuracy, it terms of allowing smaller drift-tolerances, as well as much
smoother, less glitchy playback!

Commit message and improvments by Havard Graff.

Fixes bug #640859.
 2011-11-07 11:33:32 +01:00
Felipe Contreras
3f1395afae baseaudiosink: rename some variables 2011-11-07 11:18:34 +01:00
Felipe Contreras
fbde258be6 baseaudiosink: use gst_util_uint64_scale_int when appropriate 
It's probably safer this way.
 2011-11-07 11:11:08 +01:00
Felipe Contreras
369cf3f14a baseaudiosink: split drift-tolerance into alignment-threshold 
So that drift-tolerance is used for clock slaving resync, and
alignment-threshold is for timestamp drift.
 2011-11-07 11:10:05 +01:00
Felipe Contreras
58b9818853 baseaudiosink: trivial comment fixes 
Some found by Havard Graff.

Signed-off-by: Felipe Contreras <felipe.contreras@gmail.com>
 2011-11-07 10:57:56 +01:00
Wim Taymans
2f8292b495 ringbuffer: store bpf in the right variable 2011-11-04 13:21:24 +01:00
Edward Hervey
771cbbb17c rtpbuffer: Fix compilation issues with gcc 4.6.1 2011-11-04 10:36:15 +01:00
Reynaldo H. Verdejo Pinochet
7559fb29a4 Add missing default include paths to androgenizer call 
Fixes building tag/ with Android's NDK
 2011-11-03 21:35:38 -03:00
Wim Taymans
f4bee46072 net: remove net library, it's now in core 2011-11-03 16:48:51 +01:00
Wim Taymans
a5fa136c0b update for tag API removal 2011-11-02 12:11:16 +01:00
Edward Hervey
dfc9d1658d video: Add convenience macros for accessing GstVideoInfo flags 2011-11-02 11:24:33 +01:00
Wim Taymans
4e6563d91c netbuffer: _netaddress_ -> _net_address_ 2011-11-02 09:04:28 +01:00
Wim Taymans
e2015eeb5f netaddress: updata api 2011-11-02 09:04:27 +01:00
Wim Taymans
e067e67923 rename meta* -> *meta 2011-11-02 09:04:27 +01:00
Wim Taymans
5bdfd6d899 structure: fix for api update 2011-11-02 09:04:27 +01:00
Wim Taymans
df4999aeb1 bufferlist: update for new API 2011-11-02 09:04:27 +01:00
Tim-Philipp Müller
b52c5819fb Update for pad API changes 
GstProbeType, GstProbeReturn and GstActivateMode -> GstPad*
 2011-11-01 00:34:28 +00:00
Tim-Philipp Müller
220ccdf275 audioencoder: save audio info parsed in setcaps in encoder context 
Otherwise we'll just error out when the first buffer gets pushed.
This is a porting artefact, in 0.10 the infos were allocated on the
heap, now we're doing everything with stack-allocated structs.
 2011-10-31 14:22:39 +00:00
Tim-Philipp Müller
5ee51e47a1 ext, gst, gst-libs, tests: update for tag list API changes 2011-10-31 14:22:39 +00:00
René Stadler
7eb0985282 audio: remove old C file generated from template 
Not sure how this one got pulled into a merge. In 0.10, it was moved away to
gst-template a long time ago. gstaudiofilterexample.c got generated from
gstaudiofiltertemplate.c.
 2011-10-31 15:19:54 +01:00
Wim Taymans
95281cc306 Merge branch 'master' into 0.11 2011-10-28 16:24:44 +02:00
Wim Taymans
7247eb5f2c fix compile for SEEK_TYPE_CUR removal 2011-10-28 16:11:36 +02:00
Mersad Jelacic
d430eb65c5 audiosink: avoid deadlocking audioringbuffer thread 
... when it goes into wait for ringbuffer starting just after such
having been signalled.

Fixes #661738.
 2011-10-28 14:07:40 +02:00
Wim Taymans
b70275fa10 audiofilter: use BPF for unit_size 2011-10-28 11:37:31 +02:00
René Stadler
9beff28579 audiofilter: fix get_unit_size 2011-10-28 11:24:00 +02:00
René Stadler
5d2154ff4b audiofilter: init audio info sooner 2011-10-28 11:24:00 +02:00
René Stadler
372cf41a6d audio, video: init audio/video format info to UNKNOWN format 
This is to prevent e.g. GST_AUDIO_INFO_FORMAT() from crashing on a NULL pointer
dereference when used with an unset info.
 2011-10-28 11:24:00 +02:00
Wim Taymans
01854cca80 basertppay: rename caps fields 
Make the caps fields for timestamp and seqnum match the element
properties.

See #628773
 2011-10-27 18:54:50 +02:00
Wim Taymans
9555229e79 basedepay: remove old fields 2011-10-27 18:50:32 +02:00
Wim Taymans
06311362e9 fix compilation 2011-10-27 17:26:58 +02:00
Wim Taymans
016d036137 Merge branch 'master' into 0.11 
Conflicts:
	configure.ac
	gst-libs/gst/audio/gstbaseaudiosink.c
	gst/audioconvert/channelmixtest.c
	gst/playback/gstplaybasebin.c
	gst/playback/gstsubtitleoverlay.c
	tests/examples/Makefile.am
	tests/examples/audio/Makefile.am
 2011-10-27 15:44:58 +02:00
Wim Taymans
b21bb37657 overlay: fix compilation 2011-10-27 15:29:36 +02:00
Stefan Sauer
53d7d2e966 interfaces: clean up the use of iface and class/klass 2011-10-21 14:46:48 +02:00
Mark Nauwelaerts
981070eb44 audiodecoder: having gather queue contents implies some draining is in order 
... which ensures e.g. processing and sending last fragment of reverse playback
downstream at EOS.
 2011-10-19 16:51:09 +02:00
Tim-Philipp Müller
4e59e63ff7 baseaudiosink: fix unused variable compiler warning if debugging in core is disabled 
https://bugzilla.gnome.org/show_bug.cgi?id=660150
 2011-10-19 00:32:13 +01:00
Edward Hervey
12a8fff8ac audio: Add some default channel positions 2011-10-17 12:00:55 +02:00
Edward Hervey
b4858253dc audio: Properly handle signedness in gst_audio_format_build_integer() 2011-10-17 12:00:16 +02:00
Edward Hervey
45c4a19472 audio: Indent and doc fixes 2011-10-17 11:45:39 +02:00
Edward Hervey
8268a7a20e discoverer: Only call gst_video_info_from_caps on raw video 2011-10-11 17:42:35 +02:00
Wim Taymans
f1088ed647 update for UNEXPECTED -> EOS flowreturn 2011-10-10 11:39:52 +02:00
Thiago Santos
123671bc05 libs: video: Add protection against null strings 
Check and assert if input for gst_video_format_from_string is null.
Return GST_VIDEO_FORMAT_UNKNOWN as a fallback
 2011-10-09 17:05:15 -03:00
Tim-Philipp Müller
ab949eebbd audiodecoder: update to 0.11 API after merge 2011-10-09 16:15:54 +01:00
Tim-Philipp Müller
303dbaf84b Merge remote-tracking branch 'origin/master' into 0.11 
Conflicts:
	tests/check/pipelines/vorbisdec.c
	tests/check/pipelines/vorbisenc.c
 2011-10-09 16:08:36 +01:00
Alessandro Decina
bc6f00becb audioencoder: fix compile warning 2011-10-09 16:48:18 +02:00
Mark Nauwelaerts
871b1584c9 audioencoder: only resync to upstream upon discont in perfect ts mode 
... as documented, where discont is marked here if tolerance has been
exceeded.
 2011-10-08 20:20:10 +02:00
Mark Nauwelaerts
a7ce550d04 audiodecoder: fix timestamp tolerance handling 2011-10-08 20:20:06 +02:00