mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-18 15:51:11 +00:00
Code Issues 1 Projects Releases Wiki Activity
11257 commits 44 branches 1509 tags 620 MiB
Commit graph

108 commits

Author SHA1 Message Date
Sebastian Dröge
ad42b16375 gst: Update for GST_PLUGIN_DEFINE() API change 2012-04-05 15:11:05 +02:00
Wim Taymans
25137962ad fix for caps API changes 2012-03-11 19:04:41 +01:00
Wim Taymans
642ca2bd40 audioresample: remove transform lock 
In this particular case it was not sufficient anyways because the setcaps
function didn't take the transform lock.
 2012-02-23 11:19:52 +01:00
Wim Taymans
9212619549 update for new fixate_caps function 2012-02-22 12:32:44 +01:00
Wim Taymans
fcdc385aa1 port to new map API 2012-01-25 12:30:53 +01:00
Mark Nauwelaerts
97a4f7e1e5 audioresample: fix debug message format specifier 2012-01-06 16:15:45 +01:00
Sebastian Dröge
5bdf6b3383 gst: Add new layout field to the raw audio caps 2012-01-05 10:34:25 +01:00
Wim Taymans
8a9a0bf6da audioresample: truncate in fixation 2012-01-02 15:59:09 +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
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
Kipp Cannon
4c52f4e625 audioresample: Don't emit DISCONT buffers if no discontinuity happened 
audioresample is derived from GstBaseTransform, and one of
GstBaseTransform's traits is that if the derived element does not
produce an output buffer from some input buffer then the first output
buffer after that gets flaged as a discontinuity, whether or not the
buffer actually is discontinuous from the output buffer that preceded
it. When downsampling, the audioresample element requires more than
one input sample for each output sample, and if the ratio of input to
output sample rates is high enough and the input buffers short enough
it can come to pass that the resampler does not receive enough samples
on its input to produce any output.  Currently the resampler returns
GST_BASE_TRANSFORM_FLOW_DROPPED from the transform() method in this case,
causing the next buffer to be flagged as a discontinuity. If subsequent
elements in the pipeline reset themselves on disconts, this can cause
clicks and other undesireable behaviour.

Fixes bug #665004.
 2011-11-28 18:03:22 +01:00
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
Wim Taymans
2202511e77 add parent to query function 2011-11-16 17:25:17 +01:00
Wim Taymans
372b9329b9 remove query types 2011-11-09 11:47:54 +01:00
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
Wim Taymans
dae848818d audio: rework audio caps. 
Rework the audio caps similar to the video caps. Remove
width/depth/endianness/signed fields and replace with a simple string
format and media type audio/x-raw.
Create a GstAudioInfo and some helper methods to parse caps.
Remove duplicate code from the ringbuffer and replace with audio info.
Use AudioInfo in the base audio filter class.
Port elements to new API.
 2011-08-18 19:15:03 +02:00
Wim Taymans
d679dd2c54 audioresample: fix after merge 2011-08-17 10:47:38 +02:00
Wim Taymans
33467d9629 Merge branch 'master' into 0.11 
Conflicts:
	configure.ac
	ext/pango/gsttextoverlay.c
	ext/theora/gsttheoradec.c
	gst/adder/gstadder.c
	gst/adder/gstadder.h
	gst/audioresample/gstaudioresample.c
	gst/encoding/gstencodebin.c
	gst/playback/gstdecodebin.c
	gst/playback/gstdecodebin2.c
	tests/check/elements/decodebin2.c
	tests/check/elements/playbin-compressed.c
	win32/common/libgsttag.def
 2011-08-16 18:01:14 +02:00
Vincent Penquerc'h
49ec6899f4 audioresample: fix quality setting being ignored by the resampler state 
https://bugzilla.gnome.org/show_bug.cgi?id=636562
 2011-08-12 09:55:17 +02:00
Josep Torra
5629ed74b3 Fix debug statements 
Fixes build on MacOSX

Signed-off-by: Edward Hervey <edward.hervey@collabora.co.uk>
 2011-08-10 11:15:41 +02:00
Wim Taymans
4fb67fb0da audioresample: fix for event handler change 2011-07-22 21:19:08 +02:00
Sebastian Dröge
a2162b07ad audioresample: 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.
 2011-05-27 14:31:02 +02:00
Sebastian Dröge
d8e0af1fc1 gst: Update for the GstBaseTransform::transform_caps() changes 2011-05-27 12:13:14 +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
2b9845e60f audioresample: Update for negotiation related API changes 2011-05-16 15:35:40 +02:00
Wim Taymans
94dfe80f71 -base: port to new SEGMENT API 2011-05-16 13:48:11 +02:00
Wim Taymans
c9f4e0676b -base_port to new query API 2011-05-10 18:39:07 +02:00
Wim Taymans
ec57868488 -base: don't use buffer caps 
Port to newest 0.11 core API, remove GST_PAD_CAPS and GST_BUFFER_CAPS.
 2011-05-09 13:05:12 +02:00
Wim Taymans
86a4771f8e remove buffer_alloc 2011-04-29 13:28:17 +02:00
Wim Taymans
079c152e62 Merge branch 'master' into 0.11 
Conflicts:
	gst/videoscale/gstvideoscale.c
 2011-04-25 11:20:45 +02:00
Marc Plano-Lesay
2ccd243d55 audioresample: fix unused-but-set-variable warnings with gcc 4.6 
https://bugzilla.gnome.org/show_bug.cgi?id=647294
 2011-04-24 12:43:33 +01:00
Sebastian Dröge
fc4d766e28 audioresample: Remove filter-length property, it only existed for backward compatibility 2011-04-19 11:36:35 +02:00
Sebastian Dröge
f10a8f0986 gst: Use G_DEFINE_TYPE instead of GST_BOILERPLATE 2011-04-19 11:35:53 +02:00
Wim Taymans
6e160bed3d Merge branch 'master' into 0.11 
Conflicts:
	android/alsa.mk
	android/app.mk
	android/app_plugin.mk
	android/audio.mk
	android/audioconvert.mk
	android/decodebin.mk
	android/decodebin2.mk
	android/gdp.mk
	android/interfaces.mk
	android/netbuffer.mk
	android/pbutils.mk
	android/playbin.mk
	android/queue2.mk
	android/riff.mk
	android/rtp.mk
	android/rtsp.mk
	android/sdp.mk
	android/tag.mk
	android/tcp.mk
	android/typefindfunctions.mk
	android/video.mk
 2011-04-11 11:37:51 +02:00
Havard Graff
8ff295a788 audioresample: Make src query MT-safe 
It is possible that the element might be going down while the event arrives
 2011-04-08 15:04:41 +02:00
Wim Taymans
4007076b55 Merge branch 'master' into 0.11 
Conflicts:
	ext/theora/gsttheoraenc.c
 2011-04-06 16:33:56 +02:00
Mark Nauwelaerts
5c8ed3bd47 audioresample: minor simplification 
... which avoids crashing in the off-chance that structure == NULL.
 2011-04-06 12:26:08 +02:00
Wim Taymans
3b03e23559 plugins: port some plugins to the new memory API 2011-03-27 16:35:28 +02:00
Leo Singer
5bfe1baab3 audioresample: corrected buffer duration calculation to account for nonzero initial timestamp 
Since we calculate timestamps by:

  timestamp = t0 + (out samples) / (out rate)

and durations by:

  duration = ((out samples) + (processed samples)) / (out rate) - timestamp

if t0 is nonzero, this would simplify to

  duration = t0 + (processed samples) / (out rate).

This duration is too large by the amount t0.  We should have done:

  duration = t0 + ((out samples) + (processed samples)) / (out rate) - timestamp

so that

  duration = (processed samples) / (out rate).
 2010-12-17 19:34:42 +01:00
Leo Singer
d6d2aa44ab audioresample: push half a history length, instead of a full history length, at end-of-stream so that output segment and input segment have same duration 2010-12-17 19:34:42 +01:00
Leo Singer
aac8b21678 audioresample: renamed count_gap, count_nongap to more descriptive num_gap_samples, num_nongap_samples 2010-12-17 19:34:42 +01:00
Leo Singer
6832b38527 audioresample: replaced void* with gpointer 2010-12-17 19:34:42 +01:00
Leo Singer
87f2422737 audioresample: initial filter transient discarded; unit tests passing 2010-12-17 19:34:41 +01:00
Leo Singer
b4cd3329a9 Revert "Revert "audioresample: Add GAP flag support"" 
This reverts commit 35c76b3409.

Conflicts:

	gst/audioresample/gstaudioresample.c
	gst/audioresample/gstaudioresample.h
 2010-12-17 19:34:41 +01:00
Mark Nauwelaerts
93d68ec77d audioresample: relax discont checking slightly 2010-12-13 10:10:30 +01:00
Mark Nauwelaerts
a7cf165289 audioresample: provide as much valid output ts and offset as valid input 
... by independently tracking time and offset, rather than having no offset
leading to no output ts.
 2010-12-13 10:10:15 +01:00
Stefan Kost
83c14483ed various: add a missing G_PARAM_STATIC_STRINGS flag to object properties 2010-10-13 16:13:31 +03:00
Sebastian Dröge
35c76b3409 Revert "audioresample: Add GAP flag support" 
This reverts commit 129af0d8e6.

This shouldn't be committed at all, it isn't ready and apparently
was in the wrong branch locally.
 2010-09-15 11:28:29 +02:00
Leo Singer
129af0d8e6 audioresample: Add GAP flag support 
Fixes bug #586570.
 2010-09-15 11:01:45 +02:00