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16 commits

Author SHA1 Message Date
Tristan Matthews
86f0f8b67f rtpopusdepay: assume 2 channels if sprop-stereo is missing 
Fixes #1064

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2125>
 2022-04-08 13:11:25 +00:00
Sangchul Lee
67df5815a9 rtpvp8depay: Fix crash when making 'GstRTPPacketLost' custom event 
This patch fixes a seg.fault in gst_structure_new() with warnings as below.

GLib-GObject-WARNING **:
 ../gobject/gtype.c:4330: type id '0' is invalid
GLib-GObject-WARNING **:
 can't peek value table for type '<invalid>' which is not currently referenced

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1918>
 2022-03-10 19:37:49 +00:00
Sebastian Dröge
b0afaffc5d rtp: In payloaders map the RTP marker flag to the corresponding buffer flag 
This allows downstream of a payloader to know the RTP header's marker
flag without first having to map the buffer and parse the RTP header.

Especially inside RTP header extension implementations this can be
useful to decide which packet corresponds to e.g. the last packet of a
video frame.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1776>
 2022-02-28 10:13:11 +00:00
Sanchayan Maity
cc3419daf6 rtp: ldac: Set frame count information in payload 
The RTP payload seems to be required as it carries the frame count
information. Also, gst_rtp_base_payload_allocate_output_buffer had
the second argument incorrect.

Strangely some devices like Shanling MP4 and Sony XM3 would still
work without this while some like the Sony XM4 do not.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1797>
 2022-02-26 21:09:57 +05:30
Matthew Waters
b0f72ed788 ulpfecenc: slightly safer dispose impl 
Technically dispose can be called more than once (even if gstelement is
not actually set up to do that) so need to protect against that.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1761>
 2022-02-21 09:43:33 +00:00
Matthew Waters
629b427a13 ulpfecenc: fix unmatched free() call 
One must always match a g_slice_new with a g_slice_free and a g_new with
a g_free.  This was not the case for the internal ctx struct.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1761>
 2022-02-21 09:43:33 +00:00
Matthew Waters
acc9024039 rtpulpfecenc: add some debug logging 
Like, what configuration we are using or whether a fec packet is
generated.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1761>
 2022-02-21 09:43:33 +00:00
Heinrich Kruger
6dd15acf2d rtp-hdrext-colorspace: Fix color range encoding 
The color space RTP header extension encodes color range as specified in
https://www.webmproject.org/docs/container/#Range. In other words:
0: Unspecified,
1: Broadcast Range,
2: Full range,
3: Defined by matrix coefficients and transfer characteristic.

This does not match the values of GstVideoColorRange, so it is not
correct to just write the colorimetry.range value to the header
extension.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1482>
 2021-12-30 16:31:33 +00:00
Jeongki Kim
04f6fbc237 rtpg726depay: fix endian conversion 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1469>
 2021-12-24 14:52:38 +09:00
Mathieu Duponchelle
d12d45db77 reddec: implement support for the BUNDLE case 
When multiple streams are bundled together, there may be more
than one red payload type to handle.

In addition, as the red decoder works by filling in gaps in
the seqnums, there needs to be one rtp_history queue per sequence
domain.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1429>
 2021-12-14 17:34:53 +00:00
Mathieu Duponchelle
5dc280de9f rtp/redenc|ulpfecenc: add support for TWCC 
In redenc, when input buffers have a header for the TWCC extension,
we now add one to our wrapper buffers.

In ulpfecenc we add one in that case to our protection buffers.

This makes TWCC functional when UlpRed is used in webrtcbin.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1414>
 2021-12-14 03:26:56 +00:00
Thibault Saunier
49055f1cd5 rtph264pay: Handle 'profile' field 
In order to allow "level-asymmetry-allowed" we now handle a new
"profile" field, which as the same semantics as the "profile" field in
H.264 stream so that we can force payloaded stream to have the right
format when using the `gst_sdp_media_get_caps_from_media` to set caps
filter after the payloader. This allows a simple negotiation in standard
RTP negotiation based on SDPs (like webrtc) for that particular case,
closely respecting the specs.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1410>
 2021-12-12 10:59:00 -03:00
Olivier Crête
c272d0bfcd rtopuspay: Set marker bit inside RTP packet too 
At the end of a talk spurt, not only set the marker flag on the
GstBuffer, but also set the bit inside the RTP header as recommended
by the RFC.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1124>
 2021-10-12 17:18:19 -04:00
Olivier Crête
ba328fb98d rtphdrext: Set caps without attributes as the default 
Most subclasses just use the simple function, so just let the base class
do it. It makes less code in subclasses.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/906>
 2021-09-28 20:04:55 +00:00
Olivier Crête
498740fe57 rtphdrext: Put simple caps generation as the base class default 
Instead of having a helper function that gets called by almost every
subclass, just let the base class set the caps fields automatically.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/906>
 2021-09-28 20:04:55 +00:00
Thibault Saunier
5ff769d731 Move files from gst-plugins-good into the "subprojects/gst-plugins-good/" subdir 2021-09-24 16:13:50 -03:00