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89 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
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
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
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
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
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
Tim-Philipp Müller
5f59b4f7ee Fix FSF address 
https://bugzilla.gnome.org/show_bug.cgi?id=687520
 2012-11-03 23:05:09 +00:00
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
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
248ab2d064 Fix for latest API changes 2011-03-30 16:50:45 +02:00
Wim Taymans
3b03e23559 plugins: port some plugins to the new memory API 2011-03-27 16:35:28 +02:00
Mark Nauwelaerts
d17c4c28d5 audiorate: add skip-to-first property 
API: GstAudioRate::skip-to-first
 2011-02-21 12:58:42 +01:00
Tim-Philipp Müller
4482cacb24 audiorate: use g_object_notify_by_pspec() if possible 
Use g_object_notify_by_pspec() when building against GLib >= 2.26.
This avoids the pspec lookup which takes the global paramspec pool lock.
 2010-10-07 20:54:32 +01:00
Sebastian Dröge
1c2846a0fc audiorate: Fill segment until the end on EOS 2010-09-01 11:37:37 +02:00
Edward Hervey
514a34b255 audiorate: Fix buffer offset_end when within tolerance. 
This fixes issues if we then have downstream elements that operate
on offset/offset_end.

And add the expected timestamp in the debug logs
 2010-05-26 08:51:09 +02:00
Sebastian Dröge
0a8b8ceda0 audiorate: Don't leak the input buffer in error cases 
Fixes bug #615572.
 2010-04-16 20:51:48 +02:00
Benjamin Otte
5e21fa5e0e gst_element_class_set_details => gst_element_class_set_details_simple 
Also change my email from the old university one to the current one.
 2010-03-16 17:41:50 +01:00
Mark Nauwelaerts
133e1cdb56 audiorate: correctly eat empty and dummy buffers 2009-12-26 19:20:18 +01:00
Mark Nauwelaerts
93f82f16cd audiorate: add Since marker for the new tolerance property 2009-12-21 18:50:34 +01:00
Mark Nauwelaerts
8b4f6dd43b audiorate: add a tolerance property 
It may not be uncommon for the input timestamps to experience some jitter
around the 'perfect time'.  As such, instead of regularly adding and dropping
samples, optionally allow for some tolerance in a more relaxed approach.

API: GstAudioRate:tolerance
 2009-12-15 19:51:08 +01:00
Mark Nauwelaerts
b5fe63ed79 audiorate: add documentation 2009-12-15 19:50:56 +01:00
Mark Nauwelaerts
60635a9fbc audiorate: use separate header file 2009-12-15 19:49:31 +01:00