Commit graph

137 commits

Author SHA1 Message Date
Stéphane Cerveau
a176f1a70f audiotestsrc: allow per feature registration
Split plugin into features including
dynamic types which can be indiviually
registered during a static build.

More details here:

https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1089>
2021-03-29 14:07:00 +02:00
Mathieu Duponchelle
cc516695b0 plugins: uddate gst_type_mark_as_plugin_api() calls 2020-06-06 00:41:17 +02:00
Mathieu Duponchelle
e666c9ec04 plugins: Use gst_type_mark_as_plugin_api() for all non-element plugin types 2020-06-03 22:44:23 -04:00
Thibault Saunier
b46718b1a0 audiotestsrc: Fix the way we compute EOS in reverse playback
In reverse playback we were not taking into account the current buffer
samples to check if we had reached EOS which was leading to a buffer
with PTS = CLOCK_TIME_NONE containing too many frames followed by a
useless buffer with pts=0 duration=0, and a g_critical issue in
gst_object_sync_values.

Also add a validate based test case.
Without that patch this is how the expectation fails:

``` diff
--- log-asink-sink-expected       2020-05-22 23:22:42.654384579 -0400
+++ log-asink-sink-actual  2020-05-22 23:29:35.671586380 -0400
@@ -27,5 +27,6 @@
 buffer: pts=0:00:00.058820861, due=0:00:00.023219955, flags=discont
 buffer: pts=0:00:00.035600907, due=0:00:00.023219954, flags=discont
 buffer: pts=0:00:00.012380952, due=0:00:00.023219955, flags=discont
-buffer: pts=0:00:00.000000000, due=0:00:00.012380952, flags=discont
+buffer: due=0:00:00.012380953, flags=discont
+buffer: pts=0:00:00.000000000, flags=discont
 event eos: (no structure)
 ```

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/667>
2020-05-25 08:19:02 +00:00
Niels De Graef
f91659ba92 audiotestsrc: Use G_DECLARE_FINAL_TYPE 2020-03-16 15:47:58 +00:00
Tim-Philipp Müller
289d8e53e2 Remove autotools build system 2019-10-13 14:15:43 +01:00
Thibault Saunier
909baa2360 Pass the code through codespell 2019-08-30 13:05:36 +00:00
Mathieu Duponchelle
d98835fdef doc: remove xml from comments 2019-05-30 01:12:59 +02:00
Vivia Nikolaidou
843cf07564 audiotestsrc: Max audio frequency is half the rate, not 1/4
https://en.wikipedia.org/wiki/Nyquist%E2%80%93Shannon_sampling_theorem
2019-05-17 16:35:40 +03:00
Thibault Saunier
27ba8d24ec doc: Port to hotdoc 2019-05-13 11:34:08 -04:00
Carlos Rafael Giani
c656cfb170 audiotestsrc: Improvements to the "ticks" wave
(Initially discussed in
https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/issues/305)

The ticks waveform can be useful for audio synchronization diagnostics
and other cases where the time offset between waveforms is important.
However, in its current form, it is too limited, and has problems with
discontinuities, which result in severe artifacts when this waveform
is output by a DAC.

This patch fixes some discontinuities and considerably expand the ticks
waveform's flexibility. They also introduce the notion of a "marker tick";
every Nth tick can have a different amplitude (usually one that is larger
than the others). This is useful for combining frequent oscilloscope
triggering with large time offset detection. For example, without marker
ticks, the tick intervals must not be too small, otherwise the maximum time
offset that can be unambiguously detected is quite small (for example, if
the interval is 50ms, then no time offset larger than 25ms can be
unambiguously recognized). If the tick intervals are too far apart, then
no sudden changes can be clearly observed, since the oscilloscope is not
updated quickly enough. But with marker ticks, this is not an issue: If
there's for example a tick every 100 ms, then the oscilloscope can be
triggered every 100 ms. And, if every 20th tick is a marker tick, then
time offsets of up to 1 second can be discovered, even though the time
between ticks is 100 ms.

The patch also applies some minor cleanup to the audiotestsrc documentation.
2019-01-10 16:15:47 +00:00
George Kiagiadakis
f6f8b979d6 audiotestsrc: implement producing non-interleaved audio buffers
https://bugzilla.gnome.org/show_bug.cgi?id=796739
2018-07-11 12:23:19 +03:00
Xavier Claessens
201e7c7803 Meson: Generate pc file for all plugins in base
https://bugzilla.gnome.org/show_bug.cgi?id=794568
2018-04-25 11:05:56 +01:00
Nicolas Dufresne
8e6c6266d7 Remove plugin specific static build option
Static and dynamic plugins now have the same interface. The standard
--enable-static/--enable-shared toggle are sufficient.
2017-05-16 13:42:07 -04:00
Thibault Saunier
099ac9faf2 docs: Convert gtkdoc comments to markdown
Modernizing the documentation, making it simpler to read an
modify and allowing us to possibly switch to hotdoc in the
future.
2017-03-10 18:19:17 -03:00
Carlos Rafael Giani
a257a177c3 audiotestsrc: Fix incorrect start of tick waveform
Make sure ticks start with an accumulator value of 0 by incrementing it
after filling in samples instead of before and by resetting the accumulator
every time a tick begins. This prevents it from being discontinuous at the
beginning of the tick.

https://bugzilla.gnome.org/show_bug.cgi?id=774050
2016-12-23 16:51:07 +00:00
Sebastian Dröge
031f256584 audiotestsrc: Fix segment boundary checking for reverse playback 2016-09-17 07:19:14 -04:00
Sebastian Dröge
cf18fae9de audiotestsrc: Don't adjust segment time in seek handler
basesrc already did that very well for us, adjusting it again on top of
that just breaks various non-standard seeks.
2016-09-14 16:51:30 +02:00
Nirbheek Chauhan
5c4f4ac1bd Add support for Meson as alternative/parallel build system
https://github.com/mesonbuild/meson

With contributions from:

Tim-Philipp Müller <tim@centricular.com>
Jussi Pakkanen <jpakkane@gmail.com> (original port)

Highlights of the features provided are:
* Faster builds on Linux (~40-50% faster)
* The ability to build with MSVC on Windows
* Generate Visual Studio project files
* Generate XCode project files
* Much faster builds on Windows (on-par with Linux)
* Seriously fast configure and building on embedded

... and many more. For more details see:

http://blog.nirbheek.in/2016/05/gstreamer-and-meson-new-hope.html
http://blog.nirbheek.in/2016/07/building-and-developing-gstreamer-using.html

Building with Meson should work on both Linux and Windows, but may
need a few more tweaks on other operating systems.
2016-08-20 11:09:51 +01:00
Vineeth TM
44b70ca3a1 base: use new gst_element_class_add_static_pad_template()
https://bugzilla.gnome.org/show_bug.cgi?id=763075
2016-03-24 14:25:41 +02:00
Wim Taymans
bd89f2430b audiotestsrc: increase freq limit
Raise the frequency limit and try to negotiate to a samplerate of 4*freq
when larger then the default samplerate.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=754450
2015-11-02 15:54:19 +01:00
Wim Taymans
c688eb0d88 audiotestsrc: add support for unlimited number of channels
Raise the channel limit and set the channel-mask for > 2 channels.
2015-11-02 15:46:22 +01:00
Wim Taymans
b0bf294a62 audiotestsrc: add support for all formats
Use the pack functions to also support the other audio formats we
have.
2015-11-02 13:22:18 +01:00
Tim-Philipp Müller
ec5c93f169 docs: update element example pipelines
- gst-launch -> gst-launch-1.0
- use autoaudiosink and audiovideosink more often
- review pipeline examples and descriptions
2015-05-10 11:38:19 +01:00
Tim-Philipp Müller
c680e324bc Remove obsolete Android build cruft
This is not needed any longer.
2015-04-26 18:42:34 +01:00
Luis de Bethencourt
df08f5eabe remove unused enum items PROP_LAST
This were probably added to the enums due to cargo cult programming and are
unused. Removing them.
2015-04-24 17:11:01 +01:00
Sebastian Dröge
631d356845 audiotestsrc: Report our latency properly in live mode
While we have no latency at all in theory, any other live source has the
duration of one buffer as minimum latency. Do the same in audiotestsrc.

https://bugzilla.gnome.org/show_bug.cgi?id=741879
2014-12-24 12:59:37 +01:00
Sebastian Dröge
948a4a3632 gst: Add better support for static plugins 2013-04-15 15:52:58 +02:00
Stefan Sauer
fbf2647f3e audiotestsrc: fix a comment typo from previous commit 2013-03-29 17:16:17 +01:00
Stefan Sauer
f68c95ebaa audiotestssrc: truncate the seek pos to the sample and round the time
Before it was done the other way around and that can trigger the assert that
already is in place. This also makes more sense; when seeking to time x, we want
then sample that is <= that pos.
2013-03-29 16:46:14 +01:00
Stefan Sauer
8c390fe80a audiotestsrc: simplify the caps
Drop channel-mask as we only do mon/stereo and channel-mask is optional in these
cases.
2013-03-25 16:47:02 +01:00
Simon Berg
f18d2a5a9a audiotestsrc: fix rounding errors that might cause segments to be one sample too short
https://bugzilla.gnome.org/show_bug.cgi?id=676884
2013-03-24 20:53:05 +00:00
Simon Berg
d8b42e993b audiotestsrc: fix buffer size of last buffer
The last buffer before EOS may be smaller than the maximum
size. The current code doesn't adjust for this, it only sets
the duration and offsets.

https://bugzilla.gnome.org/show_bug.cgi?id=696411
2013-03-24 20:53:05 +00:00
Tim-Philipp Müller
5f59b4f7ee Fix FSF address
https://bugzilla.gnome.org/show_bug.cgi?id=687520
2012-11-03 23:05:09 +00:00
Sebastian Dröge
3c1041d5eb Revert "gst: Add better support for static plugins"
This reverts commit d2d79e3bc2,
which was accidentially pushed.
2012-10-24 13:26:26 +02:00
Sebastian Dröge
d2d79e3bc2 gst: Add better support for static plugins 2012-10-24 12:10:44 +02:00
Mark Nauwelaerts
c629a44162 replace gst_tag_list_free with gst_tag_list_unref 2012-09-14 17:53:21 +02:00
Sebastian Dröge
99d73c94e9 tag: Update for taglist/tag event API changes 2012-07-28 00:35:02 +02:00
Wim Taymans
a2172bdb4b update for tag event change 2012-06-06 13:05:47 +02:00
Tim-Philipp Müller
3c6a3ad629 Use new gst_element_class_set_static_metadata() 2012-04-10 00:45:16 +01:00
Sebastian Dröge
ad42b16375 gst: Update for GST_PLUGIN_DEFINE() API change 2012-04-05 15:11:05 +02:00
Sebastian Dröge
65307dd132 gst: Update versioning 2012-04-04 14:55:15 +02:00
Wim Taymans
25137962ad fix for caps API changes 2012-03-11 19:04:41 +01:00
Wim Taymans
fcdc385aa1 port to new map API 2012-01-25 12:30:53 +01:00
Sebastian Dröge
dc8984d76c Merge branch 'master' into 0.11
Conflicts:
	gst-libs/gst/app/gstappsrc.c
	gst-libs/gst/audio/multichannel.h
	gst-libs/gst/video/videooverlay.c
	gst/playback/gstplaysink.c
	gst/playback/gststreamsynchronizer.c
	tests/check/Makefile.am
	win32/common/libgstvideo.def
2012-01-10 13:15:12 +01:00
Havard Graff
95be60de15 Fix various unlikely, but still potential memoryleaks in error code paths
https://bugzilla.gnome.org/show_bug.cgi?id=667311
2012-01-05 13:27:23 +00:00
Sebastian Dröge
2db0238450 audiotestsrc: Fix channel-mask handling 2012-01-05 10:34:25 +01:00
Sebastian Dröge
5bdf6b3383 gst: Add new layout field to the raw audio caps 2012-01-05 10:34:25 +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
d0bd5f04c0 update for new scheduling query 2011-11-18 17:58:58 +01:00