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Author SHA1 Message Date
Tim-Philipp Müller
454c554b11 docs: minor addition to GST_TAG_ID3V2_HEADER_SIZE docs 2011-09-12 19:55:40 +01:00
Thomas Vander Stichele
d223a6dd59 theoraenc: Fix descriptions of properties 2011-09-11 14:22:59 -04:00
Tim-Philipp Müller
55182ed841 baseaudiosrc: don't try to fixate "width" field for alaw/mulaw 
Fixes warning when trying to fixate e.g. pulsesrc ! audio/x-alaw ! fakesink.
 2011-09-10 18:30:55 +01:00
Tim-Philipp Müller
76ed3fb04d docs: fix some typos in the decodebin design document 2011-09-09 13:10:13 +01:00
Tim-Philipp Müller
0f38f86182 colorbalance: add some guards to interface methods 
https://bugzilla.gnome.org/show_bug.cgi?id=658584
 2011-09-09 13:09:43 +01:00
Vincent Penquerc'h
4095551b31 typefind: recognize Asylum modules 
Note that there is already a AMF detection for a different
magic, I'm not sure if that's a different format with the
same initials or not. AMF is used for a few different formats
(including video), so...

This fixes playbin2 playing Asylum modules.

https://bugzilla.gnome.org/show_bug.cgi?id=658514
 2011-09-09 13:54:45 +02:00
Nicolas Dufresne
25939e0218 subparse: Improve subrip type check regex 
This patch prevents timestamp like "1 1:00:00", which would have been seen
as hour 101 by our parser, and allow single digit hour, minute and seconds
as it's already supported by the parser, and also by other implementation
like in mplayer. This fixes bug 657872.

https://bugzilla.gnome.org/show_bug.cgi?id=657872
 2011-09-08 14:52:15 +02:00
Sebastian Dröge
2ad501aa51 decodebin: Update design documentation about how Parser/Converter are handled 2011-09-08 14:46:23 +02:00
Sebastian Dröge
21bc8ddcb7 Revert "Revert "decodebin2: Do a subset check before actually using a factory"" 
This reverts commit 5f5d832a3b.
 2011-09-08 14:42:13 +02:00
Sebastian Dröge
0f654f3feb Merge branch 'master' into 0.11 
Conflicts:
	docs/libs/Makefile.am
	tests/check/elements/decodebin2.c
 2011-09-08 14:42:00 +02:00
Sebastian Dröge
5f5d832a3b Revert "decodebin2: Do a subset check before actually using a factory" 
This reverts commit 50a88396ae.

See bug #658541.
 2011-09-08 13:25:27 +02:00
Sebastian Dröge
0e54d2c343 decodebin2: Don't use bufferalloc in the test elements 
This will cause not-linked errors that usually don't happen
because normal decoders/parsers will set srcpad caps before
allocating buffers from downstream.
 2011-09-07 16:44:59 +02:00
Sebastian Dröge
9e2ce5bbb5 decodebin2: Make sure to fixate Parser/Converter caps before continuing autoplugging 2011-09-07 16:44:59 +02:00
Josep Torra
a22faad18a playsink: only add text overlay if vido sink also accepts raw caps 
Fixes regression, pipeline fails with not negotiated, on media
containing subtitles when decoder/sink with custom caps is used.
 2011-09-07 16:08:38 +02:00
Sebastian Dröge
46e26824d4 decodebin2: Intersect the factory caps with the current caps for the capsfilter 
Otherwise we'll include many incompatible caps in the capsfilter that
will only slow down negotiation.
 2011-09-07 14:20:36 +02:00
Stefan Sauer
950af0438b docs: cleanup makefiles 
Remove commented out parts that we don't need. Remove "the wingo addition" - no
so useful after all. Narrow down file-globs for plugin docs.
 2011-09-07 14:14:02 +02:00
Stefan Sauer
abc96efb2a docs: add two mising enum docs 2011-09-07 14:14:02 +02:00
Sebastian Dröge
2d1dd857aa audiorate: Use complete audio caps, including the endianness field 2011-09-07 14:10:46 +02:00
Tim-Philipp Müller
f93748fbd4 decodebin2: fix element factory refcounting 
g_value_get_object() does not give us our own ref.

Fixes "Trying to dispose object "flacparse", but it still has a parent "registry0".
You need to let the parent manage the object instead of unreffing the object directly."
and similar warnings.

https://bugzilla.gnome.org/show_bug.cgi?id=658416
 2011-09-07 12:34:06 +01:00
Vincent Penquerc'h
cea0ac790f theoraenc: do not automatically override quality when using target bitrate 
If both quality and bitrate are set, libtheora will try to meet
both constraints, causing it to prefer emitting a smaller number
of good frames, to emitting the full number of frames that would
not meet the requested quality. This causes a slideshow effect
when the bitrate is low and the quality is high. And the default
theoraenc is high (48/63).

So only set quality when it is requested, and leave it unset
otherwise.

https://bugzilla.gnome.org/show_bug.cgi?id=658443
 2011-09-07 13:27:33 +02:00
Stefan Sauer
07e118ff24 Automatic update of common submodule 
From a39eb83 to 11f0cd5
 2011-09-06 21:24:33 +02:00
Christian Fredrik Kalager Schaller
f6ed715293 Add latest files to spec file 2011-09-06 19:19:43 +01:00
Stefan Sauer
284fec1d12 docs: activate overrides file to fix make distcheck 2011-09-06 20:13:30 +02:00
Tim-Philipp Müller
4529c6dc32 Merge remote-tracking branch 'origin/master' into 0.11 
Merge in doc updates for audio enums from 0.10, and get rid
of the #if #else in the enum list, since that confuses gtk-doc.

Conflicts:
	gst-libs/gst/audio/audio.c
	gst-libs/gst/audio/audio.h
 2011-09-06 16:42:42 +01:00
Wim Taymans
dc28bd1b63 audio: rename IS_LE/BE to _IS_LITTLE_ENDIAN/BIG_ENDIAN 2011-09-06 16:27:27 +01:00
Wim Taymans
f04b8fd8af audio/video add descriptions 
Add a description to the audio and video format info in case we want to use this
later.
 2011-09-06 16:46:48 +02:00
Tim-Philipp Müller
36a75bdb71 audio: update internal silent sample defines as well to match 0.11 2011-09-06 15:46:45 +01:00
Wim Taymans
c0d31dd555 rename IS_LE/BE to _IS_LITTLE_ENDIAN/BIG_ENDIAN 2011-09-06 16:46:02 +02:00
Tim-Philipp Müller
91d1112360 audio: update audio format enums to match changes in 0.11 
And add new audio format info stuff to docs.
 2011-09-06 15:36:51 +01:00
Stefan Sauer
cddf4c86d3 Automatic update of common submodule 
From 605cd9a to a39eb83
 2011-09-06 15:40:02 +02:00
Wim Taymans
8ee3da5bba Merge branch 'master' into 0.11 
Conflicts:
	gst/playback/gstsubtitleoverlay.c
	tests/check/elements/decodebin2.c
 2011-09-06 15:31:53 +02:00
Wim Taymans
7012e88090 Merge branch 'master' into 0.11 
Conflicts:
	gst-libs/gst/audio/audio.h
	gst-libs/gst/audio/gstaudiodecoder.c
	gst-libs/gst/audio/gstaudiodecoder.h
	gst-libs/gst/audio/gstaudioencoder.c
	gst-libs/gst/audio/gstbaseaudioencoder.h
	gst/playback/Makefile.am
	gst/playback/gstplaybin.c
	gst/playback/gstplaysink.c
	gst/playback/gstplaysinkvideoconvert.c
	gst/playback/gstsubtitleoverlay.c
	gst/videorate/gstvideorate.c
	gst/videoscale/gstvideoscale.c
	win32/common/libgstaudio.def
 2011-09-06 15:24:32 +02:00
Sebastian Dröge
50a88396ae decodebin2: Do a subset check before actually using a factory 
This prevents autoplugging if the caps have a non-empty intersection
but are not accepted by the next element's pad.
 2011-09-06 14:16:10 +02:00
Sebastian Dröge
c5733632ee subtitleoverlay: Use subset check instead of non-empty-intersection check to check if pads are compatible 2011-09-06 14:04:34 +02:00
Sebastian Dröge
e3530f434b playbin2: Use subset check instead of non-empty-intersection check to check if pads are compatible 2011-09-06 14:03:31 +02:00
Sebastian Dröge
4be8c44b08 decodebin2: Fix memory leak 2011-09-06 13:16:44 +02:00
Sebastian Dröge
490518cfa6 decodebin2: Add unit test for correct parser/converter negotiation 2011-09-06 13:16:44 +02:00
Sebastian Dröge
20f9d0bec5 decodebin2: Correctly negotiate format for parsers that can convert different stream formats 
This is done by adding a capsfilter after every parser/converter that contains
all possible caps supported by downstream elements. A capsfilter is necessary
here because the decoder is only selected after the parser selected a format
and the parser can't know what downstream would support otherwise.
 2011-09-06 13:16:44 +02:00
Sebastian Dröge
1df9fa9ee8 playbin2: If a audio/video sink was already selected don't check caps of all other possible sinks 2011-09-06 13:16:44 +02:00
Sebastian Dröge
a883ecfc31 decodebin2: Add Tim as author for the parser test 2011-09-06 13:16:44 +02:00
Wim Taymans
33196cdd2c audio: change audio format syntax a little 
Remove the _ in front of the endianness prefix.
Remove the _3 postfix for the 24 bits formats.
Add a _32 postfix after the formats that occupy extra space beyond their
natural size.
The result is that the GST_AUDIO_NE() macro can simply append the endianness
after all formats and that we only specify a different sample width when it is
different from the natural size of the sample. This makes things more consistent
and follows the pulseaudio conventions instead of the alsa ones.
 2011-09-06 12:06:39 +02:00
Tim-Philipp Müller
9a8a989a22 docs: more docs clean-ups 2011-09-06 10:07:33 +01:00
Vincent Penquerc'h
78f50f2d25 videorate: don't take the object lock twice in {set,get}_property 
https://bugzilla.gnome.org/show_bug.cgi?id=658294
 2011-09-06 09:44:38 +01:00
Tim-Philipp Müller
5e61db25b5 audio: fix GST_AUDIO_FORMAT_INFO_IS_*() macros to return a boolean 2011-09-05 23:28:20 +01:00
Tim-Philipp Müller
ba05716485 docs: some docs love 2011-09-05 23:28:20 +01:00
Tim-Philipp Müller
7563e0c9cf docs: add GstAudioDecoder and GstAudioEncoder to documentation 2011-09-05 23:28:20 +01:00
Tim-Philipp Müller
86e6343759 audio: rename GstBaseAudioDecoder/Encoder to GstAudioDecoder/Encoder 
API: gst_gst_audio_decoder_finish_frame()
API: gst_gst_audio_decoder_get_audio_info()
API: gst_gst_audio_decoder_get_byte_time()
API: gst_gst_audio_decoder_get_delay()
API: gst_gst_audio_decoder_get_latency()
API: gst_gst_audio_decoder_get_max_errors()
API: gst_gst_audio_decoder_get_min_latenc()y
API: gst_gst_audio_decoder_get_parse_state()
API: gst_gst_audio_decoder_get_plc()
API: gst_gst_audio_decoder_get_plc_aware()
API: gst_gst_audio_decoder_get_tolerance()
API: gst_gst_audio_decoder_get_type()
API: gst_gst_audio_decoder_set_byte_time()
API: gst_gst_audio_decoder_set_latency()
API: gst_gst_audio_decoder_set_max_errors()
API: gst_gst_audio_decoder_set_min_latency()
API: gst_gst_audio_decoder_set_plc()
API: gst_gst_audio_decoder_set_plc_aware()
API: gst_gst_audio_decoder_set_tolerance()

API: gst_gst_audio_encoder_finish_frame()
API: gst_gst_audio_encoder_get_audio_info()
API: gst_gst_audio_encoder_get_frame_max()
API: gst_gst_audio_encoder_get_frame_samples()
API: gst_gst_audio_encoder_get_hard_resync()
API: gst_gst_audio_encoder_get_latency()
API: gst_gst_audio_encoder_get_lookahead()
API: gst_gst_audio_encoder_get_mark_granule()
API: gst_gst_audio_encoder_get_perfect_timestamp()
API: gst_gst_audio_encoder_get_tolerance()
API: gst_gst_audio_encoder_get_type()
API: gst_gst_audio_encoder_proxy_getcaps()
API: gst_gst_audio_encoder_set_frame_max()
API: gst_gst_audio_encoder_set_frame_samples()
API: gst_gst_audio_encoder_set_hard_resync()
API: gst_gst_audio_encoder_set_latency()
API: gst_gst_audio_encoder_set_lookahead()
API: gst_gst_audio_encoder_set_mark_granule()
API: gst_gst_audio_encoder_set_perfect_timestamp()
API: gst_gst_audio_encoder_set_tolerance()

https://bugzilla.gnome.org/show_bug.cgi?id=642690
 2011-09-05 23:28:13 +01:00
Thiago Santos
2768ed75e0 encodebin: Select muxer further 
Sort muxers based on their caps and ranking before iterating to
find one that fits the profile.

Sorting is done by putting the elements that have a pad template
that can produce the exact caps that is on the profile. For example:
when asking for "video/quicktime, variant=iso", muxers that
have this exact caps on their pad templates will be put first on
the list than ones that have only "video/quicktime".

https://bugzilla.gnome.org/show_bug.cgi?id=651496
 2011-09-05 17:48:36 -03:00
Sebastian Dröge
de4fc848fa decodebin2: Actually iterate over the factories instead of only taking the first one 2011-09-05 20:32:42 +02:00
Stefan Sauer
81c9459771 tests: supress ERROR log output for some tests 
Be nice when we tests for correct error handling and don't spam stdout.
 2011-09-05 15:52:41 +02:00