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Author SHA1 Message Date
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
Wim Taymans
de4aeab544 video: add some more macros 2011-08-30 14:04:54 +02:00
Jan Schmidt
33d491a04f seek: Accept pipeline descriptions for audiosink/videosink 
Make the element_factory_make_or_warn utility function try parsing
the input string as a bin if element_factory_make() fails. This makes
the --audiosink/--videosink commandline options accept a pipeline
string.
 2011-08-30 18:21:31 +10:00
Jan Schmidt
105814e2c7 playsink: Try include 'pitch', if no other sink is provided 
As a default, try the pipeline 'pitch ! audioconvert ! autoaudiosink'
before trying plain autoaudiosink
 2011-08-30 18:21:31 +10:00
Wim Taymans
811a8961bf playsink: fix ts_offset refcounting 2011-08-29 13:33:49 +02:00
Wim Taymans
e694528155 base: port to 0.11 2011-08-29 13:28:08 +02:00
Wim Taymans
057aecc34e audio: fix after merge 2011-08-29 11:42:35 +02:00
Wim Taymans
dc2b00adb8 pbutils: port to new API 2011-08-29 11:38:01 +02:00
Wim Taymans
e1287b97ab Merge branch 'master' into 0.11 
Conflicts:
	ext/ogg/gstoggmux.c
	gst-libs/gst/audio/audio.c
	gst-libs/gst/audio/audio.h
	gst-libs/gst/audio/multichannel.h
	gst-libs/gst/pbutils/Makefile.am
	gst-libs/gst/pbutils/gstdiscoverer.c
	gst/playback/gstplaysinkaudioconvert.c
	gst/playback/gstplaysinkvideoconvert.c
	win32/common/libgstaudio.def
 2011-08-29 11:37:36 +02:00
Tim-Philipp Müller
67a12c9c72 pbutils: don't depend on libgstvideo just to parse some caps 
Let's extract those ints and fractions ourselves and not depend
on libgstvideo.
 2011-08-27 14:57:41 +01:00
Tim-Philipp Müller
517153e85a audio: add GstBaseAudioDecoder and GstBaseAudioEncoder to build 
However, libgstaudio now depends on libgstvideo (via pbutils).

https://bugzilla.gnome.org/show_bug.cgi?id=642690

API: gst_audio_info_clear()
API: gst_audio_info_convert()
API: gst_audio_info_copy()
API: gst_audio_info_free()
API: gst_audio_info_from_caps()
API: gst_audio_info_init()
API: gst_audio_info_to_caps()
API: gst_base_audio_decoder_finish_frame()
API: gst_base_audio_decoder_get_audio_info()
API: gst_base_audio_decoder_get_byte_time()
API: gst_base_audio_decoder_get_delay()
API: gst_base_audio_decoder_get_latency()
API: gst_base_audio_decoder_get_max_errors()
API: gst_base_audio_decoder_get_min_latency()
API: gst_base_audio_decoder_get_parse_state()
API: gst_base_audio_decoder_get_plc()
API: gst_base_audio_decoder_get_plc_aware()
API: gst_base_audio_decoder_get_tolerance()
API: gst_base_audio_decoder_get_type()
API: gst_base_audio_decoder_set_byte_time()
API: gst_base_audio_decoder_set_latency()
API: gst_base_audio_decoder_set_max_errors()
API: gst_base_audio_decoder_set_min_latency()
API: gst_base_audio_decoder_set_plc()
API: gst_base_audio_decoder_set_plc_aware()
API: gst_base_audio_decoder_set_tolerance()
API: gst_base_audio_encoder_finish_frame()
API: gst_base_audio_encoder_get_audio_info()
API: gst_base_audio_encoder_get_frame_max()
API: gst_base_audio_encoder_get_frame_samples()
API: gst_base_audio_encoder_get_hard_resync()
API: gst_base_audio_encoder_get_latency()
API: gst_base_audio_encoder_get_lookahead()
API: gst_base_audio_encoder_get_mark_granule()
API: gst_base_audio_encoder_get_perfect_timestamp()
API: gst_base_audio_encoder_get_tolerance()
API: gst_base_audio_encoder_get_type()
API: gst_base_audio_encoder_proxy_getcaps()
API: gst_base_audio_encoder_set_frame_max()
API: gst_base_audio_encoder_set_frame_samples()
API: gst_base_audio_encoder_set_hard_resync()
API: gst_base_audio_encoder_set_latency()
API: gst_base_audio_encoder_set_lookahead()
API: gst_base_audio_encoder_set_mark_granule()
API: gst_base_audio_encoder_set_perfect_timestamp()
API: gst_base_audio_encoder_set_tolerance()
 2011-08-27 14:47:50 +01:00
Tim-Philipp Müller
58f515f06a docs: add since markers to baseaudio{decoder,encoder} documentation 2011-08-27 14:47:50 +01:00
Tim-Philipp Müller
90e3d25891 baseaudiodecoder, baseaudioencoder: fix some compiler warnings 
Leaving the GST_USE_UNSTABLE_API guards in until some of the
ported decoders have been updated and it's clear that I didn't
mess up anywhere porting things to the new audio API.
 2011-08-27 14:47:49 +01:00