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Author SHA1 Message Date
Sebastian Dröge
b0b0557c48 gst: Add better support for static plugins 2013-04-15 15:54:11 +02:00
Stefan Sauer
99f84b8c4c audiopanorama: remove channel-mask from caps 
The channel-mask is only needed for channels>2 which we don't do.
 2013-02-15 21:30:15 +01:00
Stefan Sauer
86ae581928 audiopanorama: add more debug logging 2013-02-05 18:51:27 +01:00
Stefan Sauer
1f1fe47cb6 audiopanorama: further port to 1.0 
Transformcaps is not called with caps containing single structures anymore. Also add missing filter handling. Still does not negotiate though.
 2013-02-04 11:08:23 +01:00
Stefan Sauer
d187b96ee2 audiopanorama: fix caps 
We don't turn float into 32bit pcm. Looks like a typo from updating the caps.
 2013-02-03 22:45:52 +01:00
Tim-Philipp Müller
b19122bac8 scaletempo: no need for a private struct 2012-12-15 21:27:01 +00:00
Tim-Philipp Müller
61913ab7b4 audiofx: move scaletempo element from -bad 
https://bugzilla.gnome.org/show_bug.cgi?id=687262
 2012-12-14 13:16:17 +00:00
Sebastian Dröge
314765c294 scaletempo: Fix event leak 2012-12-14 13:16:17 +00:00
Sebastian Dröge
490e408991 scaletempo: Fix timestamp tracking 2012-12-14 13:16:17 +00:00
Sebastian Dröge
502eb8d1b7 scaletempo: Implement LATENCY query 2012-12-14 13:16:17 +00:00
Sebastian Dröge
c7589817cb scaletempo: Store instance private data in the instance struct 
Getting it over and over again via G_TYPE_INSTANCE_GET_PRIVATE()
is really slow.
 2012-12-14 13:16:17 +00:00
Tim-Philipp Müller
e552bd484f scaletempo: use gst_element_class_set_static_metadata() 
where possible. Avoids some string copies. Also re-indent
some stuff. Also some indent fixes here and there.
 2012-12-14 13:16:17 +00:00
Mark Nauwelaerts
d2dd91ac47 scaletempo: replace gst_element_class_set_details_simple with gst_element_class_set_metadata 2012-12-14 13:16:17 +00:00
Wim Taymans
cb1743d578 scaletempo: ffmpegcolorspace is no more 2012-12-14 13:16:17 +00:00
Sebastian Dröge
93e1091d7f scaletempo: Update for GST_PLUGIN_DEFINE() API changes 2012-12-14 13:16:17 +00:00
Mark Nauwelaerts
3286cdd542 scaletempo: port to 0.11 2012-12-14 13:16:16 +00:00
Stefan Kost
62d780cd51 scaletempo: improve the docs 
Fix the syntax, add more explanation and xref the properties.
 2012-12-14 13:16:16 +00:00
Chris E Jones
caf2b6cb5c scaletempo: Correctly handle newsegment events with stop==-1 
Fixes bug #645420.
 2012-12-14 13:16:16 +00:00
Stefan Kost
6d54058982 scaletempo: add missing G_PARAM_STATIC_STRINGS flags 
Canonicalize property names as needed.
 2012-12-14 13:16:16 +00:00
Benjamin Otte
38bc2dfb4a scaletempo: gst_element_class_set_details => gst_element_class_set_details_simple 2012-12-14 13:16:16 +00:00
Thiago Santos
2d72ec153a scaletempo: properly update new segments 
Scaletempo was missing an update of 'stop' in
new segment parameters when pushing it downstream,
which caused files to end earlier when rate < 1.

Fixes #599903

Based on patch by: Bastian Hecht <hechtb@gmail.com>
 2012-12-14 13:16:16 +00:00
Maximilian Högner
2fe7a97f1c scaletempo: Explicitely cast to signed integers to fix a segfault 
Fixes bug #585660.
 2012-12-14 13:16:16 +00:00
Michael Smith
1b1f6f56d6 scaletempo: Do not use void pointer arithmetic. 2012-12-14 13:16:16 +00:00
Stefan Kost
9284c85b33 scaletempo: Return the result of parent_class->event() 
Original commit message from CVS:
* gst/audiofx/gstscaletempo.c:
Return the result of parent_class->event().
 2012-12-14 13:16:16 +00:00
Rov Juvano
43e79f7769 Add scaletempo plugin, which allows to scale the speed of audio without changing the pitch by handling seeks with a r... 
Original commit message from CVS:
Patch by: Rov Juvano <rovjuvano at users dot sourceforge dot net>
* configure.ac:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/inspect/plugin-scaletempo.xml:
* examples/scaletempo/Makefile.am:
* examples/scaletempo/demo-gui.c: (pop_status_bar),
(status_bar_printf), (demo_gui_seek_bar_format), (update_position),
(demo_gui_seek_bar_change), (demo_gui_do_change_rate),
(demo_gui_do_set_rate), (demo_gui_do_rate_entered),
(demo_gui_do_toggle_advanced), (demo_gui_do_toggle_disabled),
(demo_gui_do_seek), (demo_gui_do_play), (demo_gui_do_pause),
(demo_gui_do_play_pause), (demo_gui_do_open_file),
(demo_gui_do_playlist_prev), (demo_gui_do_playlist_next),
(demo_gui_do_about_dialog), (demo_gui_do_quit),
(demo_gui_request_set_stride), (demo_gui_request_set_overlap),
(demo_gui_request_set_search), (demo_gui_rate_changed),
(demo_gui_playing_started), (demo_gui_playing_paused),
(demo_gui_playing_ended), (demo_gui_player_errored),
(demo_gui_stride_changed), (demo_gui_overlap_changed),
(demo_gui_search_changed), (demo_gui_set_player_func),
(demo_gui_set_playlist_func), (build_gvalue_array),
(create_action), (demo_gui_show_func), (demo_gui_set_player),
(demo_gui_set_playlist), (demo_gui_show), (demo_gui_get_property),
(demo_gui_set_property), (demo_gui_init), (demo_gui_class_init),
(demo_gui_get_type):
* examples/scaletempo/demo-gui.h:
* examples/scaletempo/demo-main.c: (handle_error_message),
(handle_quit), (main):
* examples/scaletempo/demo-player.c: (no_pipeline),
(demo_player_event_listener), (demo_player_state_changed_cb),
(demo_player_eos_cb), (demo_player_build_pipeline), (_set_rate),
(demo_player_scale_rate_func), (demo_player_set_rate_func),
(_set_state_and_wait), (demo_player_load_uri_func),
(demo_player_play_func), (demo_player_pause_func), (_seek_to),
(demo_player_seek_by_func), (demo_player_seek_to_func),
(demo_player_get_position_func), (demo_player_get_duration_func),
(demo_player_scale_rate), (demo_player_set_rate),
(demo_player_load_uri), (demo_player_play), (demo_player_pause),
(demo_player_seek_by), (demo_player_seek_to),
(demo_player_get_position), (demo_player_get_duration),
(demo_player_get_property), (demo_player_set_property),
(demo_player_init), (demo_player_class_init),
(demo_player_get_type):
* examples/scaletempo/demo-player.h:
* gst/audiofx/Makefile.am:
* gst/audiofx/gstscaletempo.c: (best_overlap_offset_float),
(best_overlap_offset_s16), (output_overlap_float),
(output_overlap_s16), (fill_queue), (reinit_buffers),
(gst_scaletempo_transform), (gst_scaletempo_transform_size),
(gst_scaletempo_sink_event), (gst_scaletempo_set_caps),
(gst_scaletempo_get_property), (gst_scaletempo_set_property),
(gst_scaletempo_base_init), (gst_scaletempo_class_init),
(gst_scaletempo_init):
* gst/audiofx/gstscaletempo.h:
* gst/audiofx/gstscaletempoplugin.c: (plugin_init):
Add scaletempo plugin, which allows to scale the speed of audio without
changing the pitch by handling seeks with a rate!=1.0.
Integrate it into the docs and add the example application for it.
Fixes bug #537700.
 2012-12-14 13:16:15 +00:00
Tim-Philipp Müller
230cf41cc9 Fix FSF address 
https://bugzilla.gnome.org/show_bug.cgi?id=687520
 2012-11-04 00:07:18 +00:00
Tim-Philipp Müller
4bb52bbadf docs: gst-launch -> gst-launch-1.0 and ffmpegcolorspace -> videoconvert 2012-08-27 21:20:30 +01:00
Tim-Philipp Müller
0fa3992e37 audiopanorama: fix negotiation and unit test 
Must remove a possibly-fixed channel-mask field if
we're going to set unfixed channels on the structure,
or a different channel count.
 2012-07-03 17:54:22 +01:00
Chris Pankow
6042bb1e6b audiofxbasefirfilter: Fix time-domain convolution for multichannel input 
Fixes bug #674025.
 2012-04-23 10:08:59 +02:00
Tim-Philipp Müller
e09ae5736d Use new gst_element_class_set_static_metadata() 2012-04-10 00:51:41 +01:00
Sebastian Dröge
aa2cd462da gst: Update for GST_PLUGIN_DEFINE() API changes 2012-04-05 17:36:38 +02:00
Sebastian Dröge
5cdd49bf25 gst: Update versioning 2012-04-04 14:37:47 +02:00
Wim Taymans
ff58bf3db9 use transform_ip_on_passthrough 2012-04-02 11:13:09 +02:00
Mark Nauwelaerts
62d6c00ac9 audiopanorama: fix supported template caps and sample processing 2012-03-29 17:21:50 +02:00
Mark Nauwelaerts
8742a0a89b audiofx: more adjustment to changed semantics of audiofilter _setup method 2012-03-28 12:23:56 +02:00
Mark Nauwelaerts
9041a588f9 audiofx: adjust to changed semantics of audiofilter _setup method 
... in that it will now call subclass with info on proposed audio format
without having set that info already in base class.  As such,
subclass can not rely on audio format info being available there.
 2012-03-23 18:48:53 +01:00
Sebastian Dröge
78bb66902b gst: Update for the gstmarshal.[ch] removal 2012-03-02 11:17:33 +01:00
Mark Nauwelaerts
f189f62b13 Merge branch 'master' into 0.11 
Conflicts:
	ext/wavpack/gstwavpackenc.c
	tests/check/elements/audioiirfilter.c
	tests/examples/v4l2/probe.c
 2012-03-01 11:29:50 +01:00
Edward Hervey
9beda57c3a Suppress deprecation warnings in selected files, for g_value_array_* mostly 2012-02-27 14:47:25 +01:00
Wim Taymans
3c292543bc audiofx: remove transform lock usage 2012-02-23 12:03:24 +01:00
Wim Taymans
44d369211c audiodynamic: fix negotiation 2012-02-06 13:28:55 +01:00
Tim-Philipp Müller
0f3b7b010e build: ignore GValueArray deprecation warnings for the time being 
until this gets sorted out with the GLib folks and we have a
viable alternative.

https://bugzilla.gnome.org/show_bug.cgi?id=667228
 2012-02-01 16:40:51 +00:00
Wim Taymans
583d39dd8d update for new memory API 2012-01-25 12:30:28 +01:00
Tim-Philipp Müller
37409d4d65 Don't use deprecated GLib API 2012-01-22 23:32:51 +00:00
Leo Singer
56353e24d2 audiofx: Use most common convention for definitions of IIR filter coefficients. 
Most signal processing texts, including MATLAB, use the following convention for IIR filter coefficients:

a_0 y[n] + a_1 y[n-1] + ... + a_M y[n-M] = b_0 x[n] + b_1 x[n-1] + ... + b[N] x[n-N]

Usually, a_0 is set to 1 because the coefficients can always be rescaled, giving

y[n] = b_0 x[n] + b_1 x[n-1] + ... + b[N] x[n-N] - a_1 y[n-1] - ... - a_M y[n-M]

The convention that was previously used by audiofxbaseiirfilter and derived class had the a and b coefficients swapped, and did not have the minus signs.

This change makes the audiofx plugin use the more common convention described above.
 2012-01-11 15:24:00 +01:00
Sebastian Dröge
93e3ed5a86 Merge branch 'master' into 0.11 
Conflicts:
	ext/cairo/gsttextoverlay.c
	ext/pulse/pulseaudiosink.c
	gst/audioparsers/gstaacparse.c
	gst/avi/gstavimux.c
	gst/flv/gstflvmux.c
	gst/interleave/interleave.c
	gst/isomp4/gstqtmux.c
	gst/matroska/matroska-demux.c
	gst/matroska/matroska-mux.c
	gst/matroska/matroska-mux.h
	gst/matroska/matroska-read-common.c
	gst/multifile/gstmultifilesink.c
	gst/multipart/multipartmux.c
	gst/shapewipe/gstshapewipe.c
	gst/smpte/gstsmpte.c
	gst/udp/gstmultiudpsink.c
	gst/videobox/gstvideobox.c
	gst/videocrop/gstaspectratiocrop.c
	gst/videomixer/videomixer.c
	gst/videomixer/videomixer2.c
	gst/wavparse/gstwavparse.c
	po/ja.po
	po/lv.po
	po/sr.po
	tests/check/Makefile.am
	tests/check/elements/qtmux.c
	tests/check/elements/rgvolume.c
 2012-01-10 14:32:32 +01:00
Sebastian Dröge
686698bf72 audiofx: Port to the new multichannel caps and the new raw audio layout field 2012-01-05 10:30:31 +01:00
Tim-Philipp Müller
66f6e12888 Work around deprecated thread API in glib master 
Add private replacements for deprecated functions such as
g_mutex_new(), g_mutex_free(), g_cond_new() etc., mostly
to avoid the deprecation warnings. We'll change these
over to the new API once we depend on glib >= 2.32.
 2011-12-12 09:46:27 +00:00
Vincent Penquerc'h
c0e101e93f various: fix pad template leaks 
https://bugzilla.gnome.org/show_bug.cgi?id=662664
 2011-11-28 13:30:27 +00:00
Matej Knopp
1e5dd9e315 Fix printf format compiler warnings on OS X / 64bit 
https://bugzilla.gnome.org/show_bug.cgi?id=662615
 2011-11-22 01:28:22 +00:00
Wim Taymans
6190312214 add parent to query function 2011-11-16 17:27:13 +01:00
Stefan Sauer
9ce6c731c3 various: add missing includes 2011-11-10 23:09:23 +02:00
Wim Taymans
49658dd5b5 remove query types 2011-11-09 11:53:01 +01:00
Stefan Sauer
fb162c8eb4 controller: port to new controller location and api 2011-11-04 20:15:48 +01:00
René Stadler
3b6de2bacd audiopanorama: simplify get_unit_size 2011-10-28 21:26:33 +02:00
René Stadler
8809965204 audioecho: fix internal buffer size calculation 2011-10-28 21:22:38 +02:00
René Stadler
42f401a7eb audiofx: fix crash in process() 2011-10-28 13:08:48 +02:00
René Stadler
9b94fc3102 audiodynamic: don't set process function too early 
GstAudioInfo and GstAudioFilter have been changed so that this code doesn't
crash anymore when a property is set in NULL state.
 2011-10-28 11:25:37 +02:00
René Stadler
7dba29cbd3 audiopanorama: fix get_unit_size 2011-10-28 11:25:37 +02:00
Wim Taymans
e204c5934c -good: port to new audio caps 2011-09-06 13:16:27 +02:00
Wim Taymans
445bf71bd1 port to more audio api changes 2011-08-19 16:09:48 +02:00
Wim Taymans
90f5b31b4b port to new audio API and caps 2011-08-19 11:49:44 +02:00
Wim Taymans
984a0b54eb fixes for event handler changes 2011-07-22 21:19:45 +02:00
Wim Taymans
7ef7157986 Merge branch 'master' into 0.11 2011-06-17 18:12:50 +02:00
Stefan Kost
6c3e77964a audioecho: fix param flags 
If the parameter cannot be changed in paused&playing, it is not controlable. Set
the appropriate mutability flag instead.
 2011-06-17 03:07:09 +03:00
Wim Taymans
409f29700d -good: port some more plugins 2011-06-13 17:51:40 +02:00
Edward Hervey
8c83978d56 audiofxbasefirfilter: Buffers no longer have caps 2011-06-07 11:22:35 +02:00
Wim Taymans
b121bb0ae9 audiofx: fix pad_alloc 2011-04-29 15:46:21 +02:00
Wim Taymans
237ca1631f port some more elements to 0.11 2011-04-25 12:49:36 +02:00
Wim Taymans
4aa6ca5578 port more plugins to 0.11 2011-04-18 10:54:43 +02:00
Sebastian Dröge
6f480ad0ed audiowsinc{band,limit}: Fix check for divison by zero 2011-04-13 18:11:34 +02:00
Sebastian Dröge
de7a976531 audiowsincband: Fix range of kernel elements (lim -> lim-1) 2011-04-13 18:01:01 +02:00
Sebastian Dröge
4fd5fea2b2 audiowsinclimit: Add some more braces to make the code more readable 2011-04-13 18:00:44 +02:00
Jordi Burguet-Castell
766e437af1 audiowsinclimit: Fix range of kernel elements (lim -> lim-1) in high/low-pass filters 2011-04-13 17:57:06 +02:00
Sebastian Dröge
2575cfc4a6 audiowsincband: Add new windowing functions: gaussian, cos and hann 2011-04-13 17:52:30 +02:00
Jordi Burguet-Castell
782d6af83d audiowsinclimimt: Add new windows to high/low-pass filters: gaussian, cosine, hann 2011-04-13 17:52:30 +02:00
Thibault Saunier
b541208b77 android: Make it ready for androgenizer 
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 01:20:11 +02:00
David Schleef
7b8981766b Change M_PI to G_PI 2010-12-30 14:20:52 -08:00
Stefan Kost
d8167e3071 various (gst): add a missing G_PARAM_STATIC_STRINGS flags 2010-10-13 18:00:28 +03:00
Sebastian Dröge
711e0cc90b audioiirfilter: Fix possible NULL pointer dereference 2010-06-16 19:24:54 +02:00
Stefan Kost
43ebe8235f docs: fix xml 
The title tag belongs into the refsect2.
 2010-04-08 10:30:06 +03:00
Benjamin Otte
cccfeaa59c gst_element_class_set_details => gst_element_class_set_details_simple 2010-03-18 14:32:00 +01:00
Stefan Kost
f405f9c775 audiopanorama: move invariant check out of the inner loop 
Improves performance for simple method.
 2010-03-11 10:35:05 +02:00
Sebastian Dröge
79e720052a audiofx: Sync properties to the stream time 2010-03-09 21:03:18 +00:00
Kipp Cannon
d009678bc5 audioamplify: Allow negative amplifications 
Fixes bug #606807.
 2010-01-13 09:22:20 +01:00
Sebastian Dröge
a9a5e0c7e1 audiofxbasefirfilter: Add property for not draining the history on kernel changes 
Currently this only works if the kernel size doesn't change, in the future
it will be possible to change the kernel size too without draining
the complete history and without loosing anything.

Partially based on a patch by
Thiago Santos <thiago.sousa.santos@collabora.co.uk>
 2010-01-07 17:28:43 +01:00
Thiago Santos
173be1422c audiofxbasefirfilter: do not try to alloc really large buffers 
When nsamples_out is larger than nsamples_in, using unsigned
ints lead to a overflow and the resulting value is wrong and
way too large for allocating a buffer. Use signed integers
and returning immediatelly when that happens.
 2009-12-26 16:59:14 -03:00
Sebastian Dröge
c26ccb9722 audiowsincband: Use the same upper length limit as audiowsinclimit 2009-12-15 18:18:54 +01:00
Sebastian Dröge
7fec6843c0 audiowsinc{limit,band}: Allow much larger filter lengths now 2009-12-15 18:12:47 +01:00
Sebastian Dröge
119a6ce637 audiofxbasefirfilter: Fix frequency response calculation 2009-12-15 18:12:47 +01:00
Sebastian Dröge
8695581751 audiofxbasefirfilter: Remove dead assignments 2009-12-15 18:12:46 +01:00
Sebastian Dröge
cd2b1c1b58 audiofxbasefirfilter: Add special processing functions for Mono/Stereo 
This provides another 7% speedup for the time domain convolution and 1.5%
speedup for the FFT convolution on Mono input.

This optimization assumes that the compiler simplifies calculations
and conditions on constant numbers and unrolls loops with a constant
number of repeats.
 2009-12-15 18:12:46 +01:00
Sebastian Dröge
a3d7321c50 audiofxbasefirfilter: Add a "low-latency" mode 
This will always use time-domain convolution, which lowers the latency.
With FFT convolution it's always a multiple of the kernel length,
with time domain convolution it's only the pre-latency of the filter kernel.
 2009-12-15 18:12:46 +01:00
Sebastian Dröge
ca568ff079 audiofxbasefirfilter: Remove obsolete TODO comments 2009-12-15 18:12:46 +01:00
Sebastian Dröge
45edc1bbd8 audiofxbasefirfilter: Use samples everywhere instead of samples*channels sometimes 2009-12-15 18:12:46 +01:00
Sebastian Dröge
02960383c1 audiofxbasefirfilter: FFT convolution implementation 
This provides a great speedup, especially the relationship between kernel
length and processing size is now logarithmic instead of linear. Below a
kernel size of 32 it's a bit slower, afterwards it's much faster:

17     0.788000 -> 0.950000
33     1.208000 -> 1.146000
65     2.166000 -> 1.146000
...
4097 107.444000 -> 1.508000

For sizes smaller 32 the normal time-domain convolution is chosen,
for larger sizes the FFT convolution is automatically used.

Fixes bug #594381.
 2009-12-15 18:12:46 +01:00
Sebastian Dröge
ddafc20b28 audiofxbasefirfilter: Make most code parts independent of the processing functions and used convolution algorithm 
Only remaining part is the residue pushing, which will be fixed later.
 2009-12-15 18:12:46 +01:00
Sebastian Dröge
43576fb0cf audiofxbasefirfilter: Optimize time-domain convolution 
Remove some redundant calculations, move comparisions out of
inner loops, etc.

This makes the convolution about 3 (!) times faster but
processing time is of course still proportional to the
filter size.
 2009-12-15 18:12:46 +01:00
Sebastian Dröge
c5f955a3b6 audiofxbasefirfilter: Use _CAST macros in some places and do some calculations only once 2009-12-15 18:12:46 +01:00
Sebastian Dröge
abb437454e audiofxbasefirfilter: Rewrite timestamp tracking 
It's much simpler now and doesn't introduce accumulating rounding
errors.
 2009-12-15 18:12:46 +01:00