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109 commits

Author SHA1 Message Date
Seungha Yang
fab2d6d60c audiorate: Update next_offset per rate change 
To support runtime audio samplerate change, re-calculate next target offset
per caps. Calculating the next buffer offset using the previous
offset seems to be tricky and rounding error prone.

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/issues/693
 2019-11-18 08:56:21 +00:00
Seungha Yang
b6bdff0c84 Revert "audiorate: accumulate offset by time diff" 
This reverts commit 4fa850e3e6.

The commit would break an constant rate audio stream with gap.
 2019-11-18 08:56:21 +00:00
Tim-Philipp Müller
289d8e53e2 Remove autotools build system 2019-10-13 14:15:43 +01:00
Thibault Saunier
27ba8d24ec doc: Port to hotdoc 2019-05-13 11:34:08 -04:00
Justin Kim
4fa850e3e6 audiorate: accumulate offset by time diff 
The fomula, 'offset = time / rate', is correct only if
the rate is never changed. When the rate is changed,
the offset should be re-calculated based on the previous
offset.

https://bugzilla.gnome.org/show_bug.cgi?id=791269
 2018-07-19 18:19:19 -04:00
George Kiagiadakis
09171b11ac audiorate: fix support for non-interleaved buffers 
https://bugzilla.gnome.org/show_bug.cgi?id=796741
 2018-07-12 13:38:27 +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
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
Tim-Philipp Müller
ddfe7a2808 win32: remove outdated build cruft 
This hasn't been touched for generations, doesn't work,
and is just causing confusion. We also don't want to
maintain these files manually.
 2016-02-20 10:05:17 +00:00
Sebastian Dröge
641428966e audiorate: Use gst_audio_format_fill_silence() instead of memset with 0 for generating silence 
For unsigned formats, silence is not all bits 0.
 2016-01-28 13:29:39 +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
Luis de Bethencourt
69f66aff9e Rename property enums from ARG_ to PROP_ 
Property enum items should be named PROP_ for consistency and readability.
 2015-04-27 11:27:00 +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
Matej Knopp
4713694082 audiorate: Fill gap events 
https://bugzilla.gnome.org/show_bug.cgi?id=741281
 2014-12-14 12:09:12 +01:00
Chad
e397b03f35 audiorate: Use gst_util_uint64_scale_int_round() 
Using gst_util_uint64_scale_int() causes slight drift
which accumulates over time.

https://bugzilla.gnome.org/show_bug.cgi?id=741045
 2014-12-02 16:07:05 -05:00
Tim-Philipp Müller
b1ff48c1a1 docs: remove old 0.10 Since markers 
They're just confusing.
 2013-11-16 16:10:07 +00:00
Matej Knopp
2f0993a95d audiorate: clip buffer before pushing it 
https://bugzilla.gnome.org/show_bug.cgi?id=708953
 2013-09-28 11:41:07 +02:00
Matej Knopp
470531d56e audiorate: ignore GAP event 
audiorate automatically fills gaps with silence.

https://bugzilla.gnome.org/show_bug.cgi?id=705048
 2013-07-29 08:23:43 +02:00
Sebastian Dröge
948a4a3632 gst: Add better support for static plugins 2013-04-15 15:52:58 +02:00
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
Tim-Philipp Müller
6d0a4ac8d5 audiorate: default to tolerance = 40ms instead of 0 
People expect audiorate to fix things up and not make things worse
by default, so let's default to a similar tolerance as audiosinks
do. Should help with transcoding and the like, though one might
possible still want higher values then.
 2012-09-09 15:58:36 +01:00
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
6c4367f6e2 audiorate: use default event handler 
Use the default event handler for unknown events.
 2012-02-03 09:56:56 +01:00
Jason DeRose
91f8f414cd audiorate: Use the number of samples for the in and out properties as documented 2012-01-27 18:16:05 +01:00
Wim Taymans
fcdc385aa1 port to new map API 2012-01-25 12:30:53 +01:00
Tim-Philipp Müller
576bbb4fd8 Remove compatibility code cruft for old GLib versions 2012-01-18 17:22:21 +00:00
Sebastian Dröge
8cd8965e19 gst: Add new layout field to all raw audio caps 2012-01-05 10:34:25 +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
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
e302833e65 add parent to pad functions 2011-11-17 12:48:25 +01:00
Wim Taymans
ab9ffa93f5 change getcaps to query 
Add sink and src event functions in rtpbasepayload
Add query vmethod to rtpbasepayload.
 2011-11-15 18:04:16 +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
beb864bd93 -base: use caps event instead of setcapsfunction 2011-06-07 10:58:27 +02:00
Sebastian Dröge
884213b8b8 base: Update for SEGMENT event parse API changes 2011-05-18 17:23:18 +02:00
Wim Taymans
94dfe80f71 -base: port to new SEGMENT API 2011-05-16 13:48:11 +02:00
Wim Taymans
816f4e791d segment: fix for new core API 
Fix for gst_*_segment_full rename.
 2011-05-09 18:16:46 +02:00
Wim Taymans
9d594f4242 audiorate: abs_rate is removed from segment structure 2011-05-09 16:42:34 +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
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
Alessandro Decina
030f639a8e 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 07:23:21 +02:00