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33 commits

Author SHA1 Message Date
Thibault Saunier
5ff769d731 Move files from gst-plugins-good into the "subprojects/gst-plugins-good/" subdir 2021-09-24 16:13:50 -03:00
Guillaume Desmottes
5fa3325335 rtpopuspay: set MARKER flag 
Set MARKER flag on first buffer after DTX.

According to RFC 3551 section 4.1 the marker bit needs to be set on
"the first packet after a silence period during which packets have
not been transmitted contiguously".

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/967>
 2021-04-26 15:25:56 +02:00
Guillaume Desmottes
41ba8c1b00 rtpopuspay: add DTX support 
If enabled, the payloader won't transmit empty frames.

Can be tested using:
  opusenc dtx=true bitrate-type=vbr ! rtpopuspay dtx=true

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/967>
 2021-04-26 15:25:56 +02:00
Stéphane Cerveau
80f8780e92 rtp: 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-good/-/merge_requests/876>
 2021-03-29 12:45:22 +02:00
Jakub Adam
5fe0aa03eb rtpopuspay: add info regarding (non-standard) multichannel support 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/832>
 2021-02-11 07:46:04 +00:00
Jakub Adam
b9ed5c7fa0 rtpopuspay: support libwebrtc-compatible multichannel payload 
When the audio has more than 2 channels, add optional fields to output
caps from which webrtcbin can generate SDP in the syntax recognized by
"multiopus" codec present in libwebrtc [1].

e.g. for 5.1 audio:

a=rtpmap:96 multiopus/48000/6
a=fmtp:96 num_streams=4;coupled_streams=2;channel_mapping=0,4,1,2,3,5

[1] https://webrtc-review.googlesource.com/c/src/+/129768

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/832>
 2021-02-11 07:46:04 +00:00
Jakub Adam
8b4147c757 rtpopuspay: make use of gst_rtp_base_payload_set_outcaps_structure() 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/832>
 2021-02-11 07:46:04 +00:00
Havard Graff
d9aaa15a30 rtpopuspay: make depay ! pay work 
There is a use-case for a server to re-payload opus going through it.

Problem was that the payloader requires channels in the caps, but
this is not something the depayloader can parse out of the stream, meaning
caps-negotiation would fail.

Removing the requirement of channels in the template-caps fixes this.
 2020-04-03 09:04:32 +00:00
Havard Graff
690c15bd78 rtpopuspay: use baseclass allocator for buffers 
That way we get some of the meta -> rtp-extension goodies.
 2019-12-02 13:05:12 +01:00
Knut Andre Tidemann
dbd7234191 rtp: opuspay: fix memory leak in gst_rtp_opus_pay_setcaps. 
The src caps were never dereferenced, causing a memory leak.
 2019-07-22 10:33:41 +02:00
Nicolas Dufresne
2b11c62571 rtppayload: Fix VP8/VP9/OPUS dual encoding name handling 
All these were copy pasted and would lead to assertion when chained with
rtpmux. This commit rewrite the negotiation with downstream. This also
drop the fallback to ancient names if the pad is unlinked. This was
completly arbitrary decision that made no sense.

https://bugzilla.gnome.org/show_bug.cgi?id=796809
 2018-08-01 09:42:36 -04:00
Tim-Philipp Müller
f9a740b319 rtp: opus: use existing utility funcs for copying/dropping metas 
We had our own copies of those while the code was in -bad, but now
we can use the existing utility functions instead of re-implementing
them.
 2017-05-24 16:18:01 +01:00
Vineeth TM
1071309870 good: use new gst_element_class_add_static_pad_template() 
https://bugzilla.gnome.org/show_bug.cgi?id=763076
 2016-03-24 14:32:20 +02:00
Tim-Philipp Müller
fb0bc126c9 rtp: opus: move Opus RTP payloader/depayloader from -bad to -good 
https://bugzilla.gnome.org/show_bug.cgi?id=756282
 2016-02-25 22:45:16 +00:00
Sebastian Dröge
01342378b5 opus: Add proper support for multichannel audio 
https://bugzilla.gnome.org/show_bug.cgi?id=757152
 2016-02-17 14:58:01 +00:00
Sebastian Dröge
0472d9f8b2 opus: 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
 2016-02-17 14:58:01 +00:00
Olivier Crête
89b172b3ed rtpopuspay: Forward stereo preferences from caps upstream 
https://bugzilla.gnome.org/show_bug.cgi?id=746617
 2016-02-17 14:58:01 +00:00
Olivier Crête
4df223f325 rtpopuspay: Set the number of channels to 2 as per RFC draft 
https://bugzilla.gnome.org/show_bug.cgi?id=746617
 2016-02-17 14:58:01 +00:00
Sebastian Dröge
bbb1143ca3 opus: Handle sprop-stereo and sprop-maxcapturerate RTP caps fields 
https://bugzilla.gnome.org/show_bug.cgi?id=746617
 2016-02-17 14:58:01 +00:00
Vincent Penquerc'h
4b5ad70924 rtpopuspay: default encoding name to OPUS 
https://bugzilla.gnome.org/show_bug.cgi?id=737810
 2016-02-17 14:58:01 +00:00
Vincent Penquerc'h
755289ed0c rtpopuspay: make caps writable before truncating them 
https://bugzilla.gnome.org/show_bug.cgi?id=737810
 2016-02-17 14:58:01 +00:00
Vincent Penquerc'h
e427369840 rtpopuspay: negotiate the encoding name 
Chrome uses a different encoding name that gstreamer.

https://bugzilla.gnome.org/show_bug.cgi?id=737810
 2016-02-17 14:58:01 +00:00
Nicolas Dufresne
9e4511edf4 rtpopus: Use OPUS encoding name 
Both Firefox and Chrome uses OPUS as the encoding in their SDP.
Adding this now defacto standard name remove the need for special
case in SDP parsing code.

https://bugzilla.gnome.org/show_bug.cgi?id=737810
 2016-02-17 14:58:01 +00:00
Wim Taymans
b310393916 opuspay: fix timestamps 
Copy timestamps to payloaded buffer.
Avoid input buffer memory leak.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692929
 2016-02-17 14:58:00 +00:00
Tim-Philipp Müller
117e30c47e Fix FSF address 
https://bugzilla.gnome.org/show_bug.cgi?id=687520
 2016-02-17 14:58:00 +00:00
Wim Taymans
5d893c7ea2 opuspay: remove pointless caps serialization 
Remove the caps serialization in the rtp caps. the spec nor the receiver
does anything with it.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686547
 2016-02-17 14:58:00 +00:00
Tim-Philipp Müller
17742d2347 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.
 2016-02-17 14:58:00 +00:00
Olivier Crête
18638c9c4e rtpopuspay: Allocate the rtp buffer correctly 
Use the right functions to allocate the rtp buffer
 2016-02-17 14:58:00 +00:00
Mark Nauwelaerts
ad261f64c3 replace gst_element_class_set_details_simple with gst_element_class_set_metadata 2016-02-17 14:58:00 +00:00
Mark Nauwelaerts
d196562755 opus: port to updated 0.11 2016-02-17 14:58:00 +00:00
Edward Hervey
77ea437507 Merge remote-tracking branch 'origin/master' into 0.11-premerge 
Conflicts:
	docs/libs/Makefile.am
	ext/kate/gstkatetiger.c
	ext/opus/gstopusdec.c
	ext/xvid/gstxvidenc.c
	gst-libs/gst/basecamerabinsrc/Makefile.am
	gst-libs/gst/basecamerabinsrc/gstbasecamerasrc.c
	gst-libs/gst/basecamerabinsrc/gstbasecamerasrc.h
	gst-libs/gst/video/gstbasevideocodec.c
	gst-libs/gst/video/gstbasevideocodec.h
	gst-libs/gst/video/gstbasevideodecoder.c
	gst-libs/gst/video/gstbasevideoencoder.c
	gst/asfmux/gstasfmux.c
	gst/audiovisualizers/gstwavescope.c
	gst/camerabin2/gstcamerabin2.c
	gst/debugutils/gstcompare.c
	gst/frei0r/gstfrei0rmixer.c
	gst/mpegpsmux/mpegpsmux.c
	gst/mpegtsmux/mpegtsmux.c
	gst/mxf/mxfmux.c
	gst/videomeasure/gstvideomeasure_ssim.c
	gst/videoparsers/gsth264parse.c
	gst/videoparsers/gstmpeg4videoparse.c
 2016-02-17 14:58:00 +00:00
Vincent Penquerc'h
8df374108a opusenc: add upstream negotiation for multistream ability 
This will help elements that cannot deal with multistream,
such as the RTP payloader.

The caps now do not include a "streams" field anymore, but
a "multistream" boolean, since we have no real use for knowing
the exact amount of streams.

https://bugzilla.gnome.org/show_bug.cgi?id=665078
 2016-02-17 14:58:00 +00:00
Danilo Cesar Lemes de Paula
c207bdf1e7 Adding opus RTP payloader/depayloader element 
Adding OPUS RTP module based on the current draft:
http://tools.ietf.org/id/draft-spittka-payload-rtp-opus-00.txt

https://bugzilla.gnome.org/show_bug.cgi?id=664817
 2016-02-17 14:58:00 +00:00