Commit graph

20 commits

Author SHA1 Message Date
Jordan Petridis
515ada7e22
Run gst-indent through the files
This is required before we enabled an indent test in the CI.

https://gitlab.freedesktop.org/gstreamer/gstreamer-project/issues/33
2018-11-28 05:52:16 +02:00
Youness Alaoui
593615de46 rtpg722pay: Add encoding-params to the src caps template
The G722 payload only accepts G722 audio with channels=1, so it must
specify the encoding-params=1 in its src caps, otherwise it causes issues
with farstream which thinks it supports 2 channels G722 and when
confronted with a remote that has G722/8000/2, it will negotiate it
and error out with a not-negotiated when the caps don't intersect
at runtime.

https://bugzilla.gnome.org/show_bug.cgi?id=789878
2017-11-03 17:20:31 -04:00
Sebastian Dröge
3010d1ec2d rtp: Filter with the filter caps in the payloader's getcaps 2016-07-25 13:35:18 +03:00
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
Hyunjun Ko
5a17572119 rtppayload: set standard payload type as default
Initialize the PT to the default value of the codec and check if
it is still the default before declaring the pt to be dynamic or
not when setting the caps.

Also use the PT constants from the rtp lib when possible

https://bugzilla.gnome.org/show_bug.cgi?id=747965
2015-08-06 01:38:43 -03:00
Tim-Philipp Müller
230cf41cc9 Fix FSF address
https://bugzilla.gnome.org/show_bug.cgi?id=687520
2012-11-04 00:07:18 +00:00
Tim-Philipp Müller
e09ae5736d Use new gst_element_class_set_static_metadata() 2012-04-10 00:51:41 +01:00
Wim Taymans
e310ee8218 caps: improve caps handling
Avoid caps copy and leaks
2012-03-27 16:42:41 +02:00
Sebastian Dröge
4885f34458 rtp: Update for the new audio caps 2012-01-05 10:30:34 +01:00
Wim Taymans
249d0083cc update for base class rename 2011-11-11 12:25:01 +01:00
Wim Taymans
fbaf216d25 update for changed base classes 2011-11-10 17:23:47 +01:00
Wim Taymans
a5cc912140 Merge branch 'master' into 0.11
Conflicts:
	ext/jpeg/gstjpegdec.c
	gst/rtp/gstrtpvrawpay.c
2011-10-13 08:58:06 +02:00
Sjoerd Simons
bf65acf11f gstrtpg722pay: Compensate for clockrate vs. samplerate difference
The RTP clock-rate used for G722 is 8000, even though the samplerate is
16000. Compensate for this by pretending G722 has 8 bits per sample
instead of the 4 bits as if it were a codec that ran at half the speed,
but with twice the number of bits. Fixes #661376
2011-10-10 21:50:28 +01:00
Wim Taymans
b0fbb1725f rtp: fix for API changes in the base classes 2011-06-13 13:25:49 +02:00
Wim Taymans
bf9b4f8362 rtp: port more to 0.11 2011-04-25 17:27:40 +02:00
Robert Swain
5b18c652fb rtp, rtpmanager: Address unused but set variables
GCC 4.6.x spits warnings about variables that are unused but set. Such
variables have been removed where trivial but with comments left behind
for informational purposes in some cases.

gst_rtp_session_chain_recv_rtcp () was changed in commit 490113d4
to always return GST_FLOW_OK instead of the return value of
rtp_session_process_rtcp (), so we'll keep it that way.
2011-04-16 12:49:16 +01:00
Wim Taymans
f4155f3cf3 rtp: add RTP hint to the klass 2010-12-21 17:23:03 +01:00
Wim Taymans
f357e09ac1 rtp: fix rank of payloaders and depayloaders
Set the payloaders and depayloaders to a reasonable rank.
2010-12-21 17:22:58 +01:00
Sebastian Dröge
c1877deee0 rtpg722pay: Fix uninitialized variable compiler warning
The clock rate is always 8000 Hz according to the RFC and
the sampling rate must always be 16000 Hz.
2010-10-03 23:49:08 +02:00
Wim Taymans
78e4a260b4 rtp: add G722 pay and depayloader 2010-09-30 18:34:36 +02:00