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
ba41bb5ca7
Merge branch 'master' into 0.11
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Conflicts:
ext/ogg/gstoggmux.c
gst/playback/gstplaysink.c
2011-08-18 19:36:50 +02:00
Wim Taymans
dae848818d
audio: rework audio caps.
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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
Vincent Penquerc'h
30236ddfd3
audioresample: fix build without orc
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https://bugzilla.gnome.org/show_bug.cgi?id=656781
2011-08-18 11:03:58 +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
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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
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https://bugzilla.gnome.org/show_bug.cgi?id=636562
2011-08-12 09:55:17 +02:00
Vincent Penquerc'h
746415a6e3
audioresample: use SSE/SSE2 when possible
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Compile in the code on i386 and x86_64, and use ORC to determine
when the runtime platform can run the code.
https://bugzilla.gnome.org/show_bug.cgi?id=636562
2011-08-12 09:55:11 +02:00
Vincent Penquerc'h
58fd202b7d
audioresample: fix SSE2 building with double precision
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The full double implementation was missing.
https://bugzilla.gnome.org/show_bug.cgi?id=636562
2011-08-12 09:53:12 +02:00
Josep Torra
5629ed74b3
Fix debug statements
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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
Wim Taymans
40d567153a
Merge branch 'master' into 0.11
2011-06-13 19:09:05 +02:00
David Schleef
4db89c82bb
convert M_PI to G_PI, for msvc
2011-06-10 23:56:34 -07:00
Wim Taymans
0ac9bb7d99
Merge branch 'master' into 0.11
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Conflicts:
tests/examples/audio/Makefile.am
tests/examples/v4l/Makefile.am
2011-06-10 12:14:57 +02:00
Tim-Philipp Müller
c692191c33
GST_PLUGINS_BASE_LIBS is not defined in -base.
2011-06-08 12:21:43 +01:00
Sebastian Dröge
a2162b07ad
audioresample: Optimize transform_caps()
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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"
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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
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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
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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
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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
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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
Alessandro Decina
030f639a8e
android: make it ready for androgenizer
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Remove the android/ top dir
Fixe the Makefile.am to be androgenized
To build gstreamer for android we are now using androgenizer which generates the
needed Android.mk files.
Androgenizer can be found here:
http://git.collabora.co.uk/?p=user/derek/androgenizer.git
2011-04-11 07:23:21 +02:00
Havard Graff
8ff295a788
audioresample: Make src query MT-safe
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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
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Conflicts:
ext/theora/gsttheoraenc.c
2011-04-06 16:33:56 +02:00
Mark Nauwelaerts
5c8ed3bd47
audioresample: minor simplification
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... 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
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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
25a154be5f
audioresample: changed num_gap_samples, num_nongap_samples from guint32 to guint64 so that gaps of greater than or equal to 2^32 samples do not cause integer overflow
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""
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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
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... 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"
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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
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Fixes bug #586570 .
2010-09-15 11:01:45 +02:00
Tim-Philipp Müller
164a91d10d
Fix build if orc is not installed
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Orc is not a hard requirement. Things should still compile and
work without orc, but slow fallback code may be used in this
case. Fix up configure to not error out if orc is not installed
and wrap use of orc profiling in audioresample in #ifdefs.
Fixes #620136 some more.
2010-06-08 13:26:53 +01:00
David Schleef
e39e729a70
audioresample: convert from liboil to orc
2010-06-07 23:58:54 -07:00
Sebastian Dröge
6723bf429f
audioresample: Update speex resampler to latest GIT
2009-11-10 12:22:27 +01:00
Robert Swain
fc56adc2e3
audioresample: fix printf variable type
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Change printf variable type from %lu to %" G_GUINT64_FORMAT " as it
should be for guint64.
Fixes #596981
2009-10-06 22:37:00 +02:00
Sebastian Dröge
1e450f21f8
audioresample: Fix drain processing
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In case we have to convert internally don't process output length input samples
but history length input samples.
2009-08-26 09:10:18 +02:00