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49 commits

Author SHA1 Message Date
Jonas K Danielsson
749652e60c rtp: Add rtppassthroughpay element 
This elements pass RTP packets along unchanged and appear as a RTP
payloader element.

This is useful, for example when using the gstreamer-rtsp-server
library, in the case where you are receiving RTP packets from a
different source and want to serve them over RTSP. Since the
gst-rtsp-server library expect the element marked as payX to be a RTP
payloader element and assumes certain properties are available.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5204>
 2023-08-22 14:01:09 +00:00
Sebastian Dröge
09045da073 rtpgstpay: Enable hdrext aggregation 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4979>
 2023-08-15 07:12:03 +02:00
Jochen Henneberg
a97d3acb90 rtp/vp8depay+vp9depay: Enable hdrext aggregation for VP8 and VP9 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4979>
 2023-08-15 07:12:03 +02:00
Jochen Henneberg
2673a66e60 rtp/h264depay+h265depay: Enable hdrext aggregation for H264 and H265 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4979>
 2023-08-15 07:12:03 +02:00
Peter Stensson
33fb3bfd60 rtpvp9pay: Only mark first outgoing packet as non delta-unit 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4937>
 2023-06-29 09:48:41 +00:00
Peter Stensson
af43648bdf rtpvp8pay: Only mark first outgoing packet as non delta-unit 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4937>
 2023-06-29 09:48:41 +00:00
Peter Stensson
b40b4ffb81 rtph265pay: Only mark first NAL as non delta-unit 
When the input buffer contained multiple NAL's the second one would keep
the non delta-unit flag for a key frame.

The delta-unit flag will now be set per NAL when preparing the buffer
list to payload.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4937>
 2023-06-29 09:48:41 +00:00
Camilo Celis Guzman
0cee3cd833 rtpvp8pay: rtpvp9pay: access picture_id property atomically 
Atomically set and get the picture_id. This changeset only atomically gets
the picture-id when such property is queried on the element, on every other
place where it is accessed internally it is accessed directly.

This is because there is no MT scenario where we would be modifying this value
and reading it internally in parallel.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4530>
 2023-05-05 07:45:19 +00:00
Camilo Celis Guzman
e4d8cda9a1 rtpvp8pay, rtpvp9pay: increment PictureID on FLUSH_START 
In recent versions of Chrome (M106) a change on their jitter buffer means that
they are very susceptible to PictureID discontinuities.

Then avoid at all cost resetting the PictureID. Moreover, according to
the RFCs for VP8 and VP9 payloads; the PictureID can start off at any
random value. So there is no logical problem of incrementing it here
rather than resetting it, as long as it is a different PictureID.

WebRTC's recent corruption issue:
https://bugs.chromium.org/p/webrtc/issues/detail?id=15101

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4530>
 2023-05-05 07:45:19 +00:00
Camilo Celis Guzman
f159fd8568 rtpvp8pay, rtpvp9pay: expose picture-id as a property 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4530>
 2023-05-05 07:45:19 +00:00
Camilo Celis Guzman
11187a81c3 rtpvp9pay: add picture-id-offset property 
Bring the VP9 payloader in sync in this regard to the VP8 payloader

Allowing setting the picture id to a known value is useful when testing.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4530>
 2023-05-05 07:45:19 +00:00
Camilo Celis Guzman
7cffb40c2e rtpvp9pay: minor refactor of PictureID logic 
This brings the logic inline with the vp8pay

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4530>
 2023-05-05 07:45:19 +00:00
Camilo Celis Guzman
a79616ea7a rtpvp8pay: avoid reseting PictureID if NO_PICTURE_ID mode is set 
There is no logical change here, as `& (1 << nbits) - 1` would produce also 0
when NO_PICTURE_ID mode is choosen. However, this avoid computing a random
integer that is actually unused.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4530>
 2023-05-05 07:45:19 +00:00
Camilo Celis Guzman
7dd6375c5e rtpvp8pay, rtpvp9pay: use GType like name for PictureIDMode 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4530>
 2023-05-05 07:45:19 +00:00
Tim-Philipp Müller
ba417b0e07 rtpjpegdepay: fix logic error when checking if an EOI is present 
We wouldn't add the missing EOI marker if the frame ended with
either 0xFF NN or 0xNN D9.

Fixes #2407

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4256>
 2023-03-24 19:39:33 +00:00
Edward Hervey
ee759fb4bf plugins: Fix wrong enum usage 
gcc 13 now detects conflicting enum usages. Fix the various cases where it was wrong

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4225>
 2023-03-20 11:40:30 +00:00
Tim-Philipp Müller
0fc568c6b1 gst-plugins-good: re-indent with GNU indent 2.2.12 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4182>
 2023-03-17 03:18:54 +00:00
Arun Raghavan
0ed51294e0 rtpopusdepay: Assume 48 kHz if sprop-maxcapturerate is missing 
This matches 7587, section 6.1:

>   sprop-maxcapturerate:  a hint about the maximum input sampling rate
>      [...]
>      bandwidths (Table 1).  By default, the sender is assumed to have
>      no limitations, i.e., 48000.

Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2354
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4151>
 2023-03-14 11:09:08 -04:00
Patricia Muscalu
c3e52d5c4f rtph264pay: Don't insert SPS/PPS before the second image slice 
Only the first slice, for which fist_mb_in_slice is set to 0,
should trigger insertion of SPS and PPS buffers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3402>
 2023-02-08 12:10:11 +00:00
Mathieu Duponchelle
2048a0a4d9 redenc: fix setting of extension ID for twcc 
1 was previously hardcoded in, and the bug went under the radar because
webrtcsink hardcodes the number too.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3785>
 2023-01-24 22:52:48 +00:00
Tim-Philipp Müller
e66f8cff26 rtp: drop use of GSlice 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3695>
 2023-01-24 15:25:06 +00:00
Olivier Crête
c593930055 rtopuspay: Use GstStaticCaps to cache parsed caps 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3674>
 2023-01-12 18:48:35 -05:00
Olivier Crête
46a6f72f03 rtopuspay: Ignore the stereo parameter in multiopus caps 
Also add unit tests for the various variants

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3674>
 2023-01-12 18:48:35 -05:00
Olivier Crête
f1cf457811 rtpopuspay: Leave original caps as-is 
This should make it work if someone specifies stereo with MULTIOPUS
somehow.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3674>
 2023-01-12 18:48:35 -05:00
Olivier Crête
c52c66b575 rtpopuspay: Return upstream channel filter based on OPUS vs MULTICAPS 
Only allow 1 or 2 channels if the caps are OPUS, or 3+ if they are
MULTIOPUS.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3674>
 2023-01-12 18:48:35 -05:00
Olivier Crête
c51ae6112d rtpopus: Put MULTIOPUS in all caps 
The RTP payload encoding-name are always in caps in GStreamer.
In SDP, they are not case-sensitive, but since caps are, we need to pick
a caps, and we picked upper-case along time ago.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3674>
 2023-01-12 18:48:35 -05:00
Mathieu Duponchelle
fa71217502 rtpvp9depay: expose keyframe-related properties 
This simply brings in the wait-for-keyframe and request-keyframe
properties from rtpvp8depay.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/909>
 2022-12-10 13:28:07 +00:00
Matthew Waters
093e9c8c9d rtpulpfecdec: add property for passthrough 
Support for enabling and disabling decoding of FEC data decoding on
packet loss events and unconditional seqnum rewriting of packets.

See
https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/581
for background.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3212>
 2022-10-23 23:44:07 +00:00
Patricia Muscalu
3c9e4f4886 rtph265: keep delta unit flag 
Without this patch all buffers that pass the payloader
are marked as non-delta-unit buffers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2969>
 2022-09-02 08:56:13 +00:00
Thibault Saunier
6a4425e46a meson: Call pkgconfig.generate in the loop where we declare plugins dependencies 
Removing some copy pasted code

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2970>
 2022-09-01 21:17:35 +00:00
Sebastian Dröge
cbc6761199 rtpvp8depay: If configured to wait for keyframes after packet loss, also do that if incomplete frames are detected 
This can happen if the data inside the packets is incomplete without the
seqnums being discontinuous because of ULPFEC being used.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2947>
 2022-08-31 08:58:03 +00:00
Sebastian Dröge
ed425e2785 rtpgstpay: Don't push packets before the first input buffer is received 
It's not possible to create a valid RTP timestamp for them, which would
cause a potentially very big RTP timestamp discontinuity between those
first packets (created from initial events) and the packet based on the
first input buffer.

As a side-effect, also simplify the packet aggregation code a bit and
work with only a single level of buffer lists.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1157

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2250>
 2022-04-27 11:55:17 +00:00
Mathieu Duponchelle
3391a7d499 rtpredenc: quieten warning about ignoring header extensions 
Turn it into a FIXME, and only log once

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2279>
 2022-04-23 01:04:54 +00:00
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