Commit graph

10161 commits

Author SHA1 Message Date
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
Tim-Philipp Müller
b1c00adf31 Revert "playsink: Try include 'pitch', if no other sink is provided"
This reverts commit 105814e2c7.

The general consensus seems to be that we should revert this for
now. If such behaviour is desired, we should probably enable it
via a flag. And maybe use the scaletempo plugin instead.
2011-09-05 14:44:27 +01:00
Sebastian Dröge
705ca1d55a playsink: Don't leak the videochain ts-offset element
Also don't leak the audiochain ts-offset element if one is
found but the sink doesn't support volume settings.
2011-09-05 12:02:37 +02:00
Sebastian Dröge
89a899fd9d playsink: Use gst_object_unref() instead of g_object_unref() for better debugging 2011-09-05 11:55:59 +02:00
David Schleef
4e38577b30 videoscale: Add modified Lanczos scaling method
Adds a Lanczos-derived scaling method, which is rather slow, but very
high quality.  Adds a few properties that can be used to tune various
scaling properties: sharpness, sharpen, envelope, dither.  Not currently
Orcified, but was designed with that in mind.
2011-09-01 15:16:50 -07:00
David Schleef
924f743981 playback: Add define for colorspace element
Single point of change if you want to switch from ffmpegcolorspace
to colorspace.
2011-09-01 11:41:31 -07:00
Sjoerd Simons
08ac05a06c videorate: fix dynamically changing average period
The average_period_set variable can be accessed in different threads, so
always lock it when reading. Furthermore when switching to averaging
mode we should make sure we don't have cached buffers that aren't used
in that mode. And any modeswitch will cause the latency to change, so we
should post a NewLatency message
2011-08-31 14:13:56 +01:00
Sjoerd Simons
ea46b3c706 videorate: Port to basetransform 2011-08-31 14:13:56 +01:00
Sjoerd Simons
f2438913f9 Correct added versions 2011-08-31 14:13:55 +01:00
Sebastian Dröge
49b301bcd6 playsink: Only unref ts_offset elements if they're not NULL 2011-08-31 14:45:08 +02:00
Wim Taymans
2f2aa4ac32 video: improve docs a little 2011-08-31 13:32:21 +02:00
Sebastian Dröge
425d3ae7bc decodebin2: Keep the chain mutex locked while connecting to the notify::caps signal 2011-08-31 12:40:30 +02:00