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22 commits

Author SHA1 Message Date
Thibault Saunier
2fd28195ca Move files from gst-plugins-base into the "subprojects/gst-plugins-base/" subdir 2021-09-24 16:13:26 -03:00
Havard Graff
0826fb95b7 audio: video: Optimize by using cached quark for meta tag 
Avoid taking the global quark lock for every single buffer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/295>
 2020-06-27 09:23:10 +00:00
Thibault Saunier
287897e465 doc: Fix some gtk-doc comments 2019-05-13 11:34:08 -04:00
Stian Selnes
f766b85b96 rtpbasepayload: rtpbasedepayload: Add source-info property 
Add a source-info property that will read/write meta to the buffers
about RTP source information. The GstRTPSourceMeta can be used to
transport information about the origin of a buffer, e.g. the sources
that is included in a mixed audio buffer.

A new function gst_rtp_base_payload_allocate_output_buffer() is added
for payloaders to use to allocate the output RTP buffer with the correct
number of CSRCs according to the meta and fill it.

RTPSourceMeta does not make sense on RTP buffers since the information
is in the RTP header. So the payloader will strip the meta from the
output buffer.

https://bugzilla.gnome.org/show_bug.cgi?id=761947
 2018-10-10 14:38:01 -04:00
Tim-Philipp Müller
7f9730ecf4 rtp: Update for g_type_class_add_private() deprecation in recent GLib 
https://gitlab.gnome.org/GNOME/glib/merge_requests/7
 2018-06-23 22:22:22 +02:00
Mark Nauwelaerts
fe6f65b1c2 rtp: fix some GIR annotations 
Mostly related to out and array parameters.
 2018-04-23 19:33:12 +02:00
Sebastian Dröge
a3656ba29b libs: Check if meta transform_func is NULL before using it 
https://bugzilla.gnome.org/show_bug.cgi?id=782050
 2017-05-02 14:31:14 +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
Sebastian Dröge
a0f1b964f1 rtpbaseaudiopayload: Copy metadata in the (de)payloader, but only the relevant ones 
The payloader didn't copy anything so far, the depayloader copied every
possible meta. Let's make it consistent and just copy all metas without
tags or with only the audio tag.

https://bugzilla.gnome.org/show_bug.cgi?id=751774
 2015-08-11 12:46:36 +02:00
Sebastian Dröge
353fb82ea1 rtpbaseaudiopayload: Don't copy memory if not needed, just append payload to the RTP buffer 2015-06-30 10:39:01 +02:00
Sebastian Dröge
abcaa71485 rtpbaseaudiopayload: Use GST_BUFFER_PTS() instead of GST_BUFFER_TIMESTAMP() 2015-06-10 14:33:01 +02:00
Olivier Crête
c6fd304eb6 rtpbaseaudiopayload: Avoid copying the data 2013-08-18 22:24:08 -04:00
Tim-Philipp Müller
71e46b2478 gst_adapter_prev_timestamp -> gst_adapter_prev_pts 
https://bugzilla.gnome.org/show_bug.cgi?id=675598
 2012-11-14 00:03:15 +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
Evan Nemerson
63579633f5 rtpbaseaudiopayload: add transfer annotation to get_adapter return 2012-07-17 11:10:04 +02:00
Edward Hervey
2817bdadc9 libs: Remove "Since" markers and minor doc fixups 2012-07-13 12:11:06 +02:00
Wim Taymans
baa2fac2f8 audiopayload: disable broken bufferlist handling 
The bufferlist handling is broken so make sure it is never enabled.
 2012-06-06 16:40:24 +02:00
Edward Hervey
ea0ed511f8 rtp: Initialize GstRTPBuffer before usage 2011-12-05 18:42:24 +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
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
107d5a3d05 rtp: fix headers 
indent, add padding, remove old abidata
 2011-11-11 19:21:09 +01:00
Wim Taymans
5f1312b5d8 rename files to match object names 2011-11-11 12:32:23 +01:00
Renamed from gst-libs/gst/rtp/gstbasertpaudiopayload.c (Browse further)