mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-10-02 16:52:42 +00:00
Code Issues 1 Projects Releases Wiki Activity
9353 commits 44 branches 1508 tags 325 MiB
Commit graph

2702 commits

Author SHA1 Message Date
Sebastian Dröge
4fcd621101 audioconvert: Use new gst_caps_is_subset_structure() API 
This prevents one copy of every structure and creating a new caps
instance.
 2011-05-27 14:10:50 +02:00
Sebastian Dröge
d590bce5f7 audioconvert: Optimize transform_caps() 
If the second and next caps structures are a subset of the already existing
transformed caps we can safely skip them because we would transform them to
the same caps again.

This makes gst_pad_get_caps() on an audiotestsrc ! audioconvert !
audioconvert ! audioconvert ! fakesink pipeline about 1.7 times faster.
 2011-05-27 13:13:42 +02:00
Sebastian Dröge
d8e0af1fc1 gst: Update for the GstBaseTransform::transform_caps() changes 2011-05-27 12:13:14 +02:00
Wim Taymans
8501753033 -base: change for changed set_blocked API 2011-05-26 16:18:16 +02:00
Sebastian Dröge
bf08ca7020 Merge branch 'master' into 0.11 2011-05-26 13:54:09 +02:00
Sebastian Dröge
216258fbc2 playbin2: Let the input-selectors sync all streams to the running time 
This is especially needed when switching between a non-sparse and sparse
video stream, see bug #537382. It also lowers the time needed for switching
between streams a bit.
 2011-05-26 11:41:50 +02:00
Stefan Kost
5cd0e0f666 audiotestsrc: add blue and violet noise by using spectral inversion 
Add blue and violet noise by spectral inversion of pink and red noise.
Fixes #649969
 2011-05-26 00:18:55 +03:00
Stefan Kost
1cf831e74e audiotestsrc: add red (brownian) noise generator 
Add another noise generator which produces a quite dark noise color.

Fixes parts of #649969.
 2011-05-25 23:43:56 +03:00
Stefan Kost
1916eecc30 volume: use a flag for 'mute' using the controller 
Previously we checked mute_csource to determine wheter we need to premultiply
volumes and mute values. That fails as we unrefs mute_csource and set it to
NULL after. Use an extra flag instead.
 2011-05-25 14:28:18 +03:00
Wim Taymans
e614c6bd81 feature: use object name instaed of feature name 2011-05-24 18:21:06 +02:00
Wim Taymans
010add200a scheduling: port to new scheduling query 2011-05-24 17:37:45 +02:00
Wim Taymans
a87c021237 Merge branch 'master' into 0.11 
Conflicts:
	gst-libs/gst/video/convertframe.c
 2011-05-24 09:47:15 +02:00
Sebastian Dröge
c867f677c0 subparse: Try to typefind even if conversion to UTF8 failed 
Fixes bug #600043.
 2011-05-23 16:05:35 +02:00
Sebastian Dröge
2c7cdbc2ca subparse: Compile the typefind regex with optimization to speed up matching 2011-05-23 16:05:35 +02:00
Sebastian Dröge
3dc4987138 subparse: Interprete typefind strings passed to GRegex as raw bytes instead of valid UTF8 2011-05-23 16:05:35 +02:00
Stefan Kost
762bb236fa docs: fixup appsrc/sink api docs 2011-05-23 15:02:27 +03:00
Wim Taymans
6688eea15c encodebin: use new method 
Use the new element class method to get the metadata from an element. Not all
elements might have an elementfactory.
 2011-05-20 12:50:39 +02:00
Sebastian Dröge
1be10803ba encodebin: Fix compilation after the removal of GstElementDetails 2011-05-20 12:32:35 +02:00
Sebastian Dröge
a9b134d1a9 Merge branch 'master' into 0.11 
Conflicts:
	docs/plugins/gst-plugins-base-plugins.hierarchy
	docs/plugins/gst-plugins-base-plugins.interfaces
	docs/plugins/gst-plugins-base-plugins.prerequisites
 2011-05-20 12:26:57 +02:00
Stefan Kost
f514be993c audioconvert: cleanup helper code 
make_lossless_changes() returns the same structure that we're passing (probably
to enable chaining). Instead of reusing s and making it point to s2 as well,
keep using s2. Drop the assignment which in the 2nd case is a dead one anyway.
 2011-05-19 23:41:08 +03:00
Thiago Santos
b5fb542386 encodebin: Autoplug formatters 
Autoplug formatters for streams if a formatter with secondary or
higher rank is found. Formatters are autoplugged when there is no
muxer or when the muxer doesn't implement the tagsetter interface.

Currently only the first formatter found is plugged, this might
help in lots of cases, but it doesn't solve the
'lamemp3 ! xingmux ! id3mux'
case.

https://bugzilla.gnome.org/show_bug.cgi?id=649841
 2011-05-19 08:35:46 -03:00
Thiago Santos
0060900d68 encodebin: fix typos 2011-05-19 08:30:03 -03:00
Sebastian Dröge
884213b8b8 base: Update for SEGMENT event parse API changes 2011-05-18 17:23:18 +02:00
Sebastian Dröge
a46485e357 gst: Update for caps/pad template related API changes 2011-05-17 13:06:01 +02:00
Sebastian Dröge
03f9ca3d64 encodebin: Update for caps/pad template related API changes 2011-05-17 13:06:00 +02:00
Sebastian Dröge
318ed07598 Revert "-base_port to new query API" 
This reverts commit c9f4e0676b.
 2011-05-17 11:25:31 +02:00
Sebastian Dröge
8549dd352d Revert "decodebin2: Update for GstQuery related API changes" 
This reverts commit 549128c2a3.
 2011-05-17 11:24:18 +02:00
Sebastian Dröge
07617d50d3 videorate: Update for segment API changes 2011-05-16 17:14:08 +02:00
Sebastian Dröge
549128c2a3 decodebin2: Update for GstQuery related API changes 2011-05-16 17:13:14 +02:00
Sebastian Dröge
34849f0108 playsink: Update for other 0.11 API changes 2011-05-16 17:13:04 +02:00
Sebastian Dröge
01c798605a playsink: Use correct number of parameters to gst_pad_get_caps() 2011-05-16 17:08:45 +02:00
Sebastian Dröge
d0362c2b87 Merge branch 'master' into 0.11 
Conflicts:
	configure.ac
	ext/alsa/gstalsasrc.c
	gst-libs/gst/audio/gstbaseaudiosink.c
	gst-libs/gst/tag/gstxmptag.c
	gst/playback/gstsubtitleoverlay.c
	gst/videorate/gstvideorate.c
	sys/xvimage/xvimagesink.c
 2011-05-16 17:06:22 +02:00
Sebastian Dröge
e21ec369f0 videorate: Update for negotiation related API changes 2011-05-16 15:35:41 +02:00
Sebastian Dröge
7330318e2f tcpclientsrc: Update for negotation related API changes 2011-05-16 15:35:41 +02:00
Sebastian Dröge
2b9845e60f audioresample: Update for negotiation related API changes 2011-05-16 15:35:40 +02:00
Sebastian Dröge
616181901e playback: Update for negotiation related API changes 2011-05-16 15:35:40 +02:00
Sebastian Dröge
a78bf729a9 ffmpegcolorspace: Update for negotiation related API changes 2011-05-16 15:35:40 +02:00
Sebastian Dröge
709b831a72 encodebin: Update for negotiation related API changes 2011-05-16 15:35:40 +02:00
Sebastian Dröge
c020add91e audioconvert: Update for negotiation related API changes 2011-05-16 15:35:40 +02:00
Sebastian Dröge
48654cdacf adder: Update for negotiation related API changes 2011-05-16 15:35:40 +02:00
Sebastian Dröge
219b610fdd videotestsrc: Update for negotiation related API changes 2011-05-16 15:35:18 +02:00
Sebastian Dröge
0d39d5bb8f videoscale: Update for negotiation related API changes 2011-05-16 15:35:17 +02:00
Sebastian Dröge
cc785bade6 videoscale: basetransform is now better at trying passthrough, remove workaround 2011-05-16 15:34:49 +02:00
Wim Taymans
94dfe80f71 -base: port to new SEGMENT API 2011-05-16 13:48:11 +02:00
Mark Nauwelaerts
eba4a948fb videorate: optionally ensure maximum average output frame rate 
See #628764.
 2011-05-16 12:46:00 +02:00
Alexey Fisher
1e09272024 videorate: optionally only drop frames to ensure maximum frame rate 
This adds option to arrange for maximal allowed variable frame rate.

Fixes #628764.
 2011-05-16 12:45:47 +02:00
Mark Nauwelaerts
4ba6acdba5 uridecodebin: use bitrate to configure streaming buffer-duration default case 
In particular, in audio only cases whose (estimated) metadata provides bitrate
information, the buffer-size based on such bitrate (and buffer-duration)
will be much more reasonable than queue2 default buffer-size.
 2011-05-16 12:44:52 +02:00
Mark Nauwelaerts
8480b1ef1d uridecodebin: remove some dead code 
... which was dead as pads were never added to the list, and need not be added,
since removing them is handled by a pad callback.
 2011-05-16 12:44:50 +02:00
Thiago Santos
fd486588ce encodebin: Check for missing converters 
Adds checks for missing video and audio converter elements
 2011-05-15 13:02:39 -03:00
Andoni Morales Alastruey
dd36e4cd0e decodebin2: fix preroll for streams at low bitrates 
For streams at low bitrates we need to set a limit in time because the limit
in bytes might not reached too late, sometimes more than 30 seconds.
This limit can only be set if upstream is seekable (see #584104)
Closes #647769
 2011-05-14 11:42:33 +02:00