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Author SHA1 Message Date
Matthew Waters
5ada3da580 tests/rtpbin2: improve performance of receive_list_benchmark 
Instead of receiving each output buffer through GstHarness individually,
process the received buffer lists directly by prociding our own pad and
chain/chain_list handlers.  Reduces test time locally from 30+s to <2s.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/2081>
 2025-02-17 09:21:15 +00:00
Sebastian Dröge
9d6272b8e0 Update to rand 0.9 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/2082>
 2025-02-17 09:26:35 +02:00
Yaakov Selkowitz
f7ba4c40a7 Add missing copies of license files 
This should fix the crates that are missing license files.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/2050>
 2025-01-27 19:26:53 -05:00
Tim-Philipp Müller
74760e1b42 rtp: ac3: validate depayloaded AC-3 data in unit tests 
Check for valid frame header.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/2044>
 2025-01-20 18:32:50 +00:00
Tim-Philipp Müller
f6d21810ff rtp: tests: add run_test_pipeline variants with data validation 
So we can actually check the content of depayloaded buffers too.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/2044>
 2025-01-20 18:32:50 +00:00
Tim-Philipp Müller
5ccef7a453 rtpac3depay2: fix handling of non-fragmented payloads 
The frames of a non-fragmented payload would contain
an extra two bytes before the frame sync and then
missing two bytes at the end which which would cause
decoding errors on the last block and/or frame crc
check failures.

This happened because we didn't take into account
the 2-byte packet payload header when creating output
sub-buffers, as the offsets we were using were in
relation to the payload data after the headers.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/645

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/2044>
 2025-01-20 18:32:50 +00:00
Sebastian Dröge
8e62e54cc9 rtp: basepay: Only forward buffers if we have a segment 
If there are pending buffers without a segment then they must come from
the caps only and should be forwarded at a later time, if any.

Also reject any incoming buffers if no segment was received.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/2037>
 2025-01-15 09:43:12 +00:00
Sebastian Dröge
536f4db5c1 rtp: basedepay: Only forward buffers if we have a segment 
If there are pending buffers without a segment then they must come from
the caps only and should be forwarded at a later time, if any.

Also reject any incoming buffers if no segment was received.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/2037>
 2025-01-15 09:43:12 +00:00
Sebastian Dröge
7b4665c793 Fix some new clippy 1.84 warnings 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/2032>
 2025-01-10 10:08:38 +02:00
Sebastian Dröge
81ff664666 rtp: Add AMR NB/WB RTP payloader/depayloader 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/2016>
 2025-01-02 16:42:14 +00:00
Sebastian Dröge
6ee745edee Update for GLib signal accumulator API changes 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1954>
 2024-11-30 15:10:06 +02:00
Sebastian Dröge
6aeb3f2af2 Fix / silence various new Rust 1.83 clippy warnings 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1954>
 2024-11-30 14:57:24 +02:00
Sebastian Dröge
347bee16d4 Update for GLib signal API changes 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1936>
 2024-11-22 15:52:41 +02:00
François Laignel
a8146f333f all: use builder conditional setters where applicable 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1926>
 2024-11-21 12:57:16 +00:00
Sebastian Dröge
ef39046e18 Update to thiserror 2 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1911>
 2024-11-06 11:02:41 +02:00
Jerome Colle
0df2b72ff2 rtpbasedepay2: fix reference timestamp meta duplicates 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1890>
 2024-10-25 18:23:07 +02:00
Sebastian Dröge
4abc5c7a48 Be stricter with Impl-trait bounds to enforce type hierarchies 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1871>
 2024-10-22 13:43:12 +00:00
Sebastian Dröge
7e59c3f0fd Remove once_cell dependency 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1868>
 2024-10-21 17:53:18 +00:00
Sebastian Dröge
120c62964d Update to bitstream-io 2.5 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1864>
 2024-10-20 19:53:15 +00:00
Matthew Waters
d4fd21d197 rtp2/jitterbuffer: check for event query earlier 
If a serialized query arrives (e.g. allocation) and the jitterbuffer has never
received a packet, then jitterbuffer would never forward the serialized query
resulting in a hang.

Fix by forwarding queries/events before the conditions that require the first
packet to arrive.

Also update unit test to check for this scenario.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1846>
 2024-10-09 16:21:13 +00:00
Sebastian Dröge
ceb88d960f rtpav1depay: Add wait-for-keyframe and request-keyframe properties 
These behave the same as the properties in other depayloaders. Keyframe
detection is based on the N flag in the aggregation header.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/598

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1823>
 2024-09-27 12:25:16 +03:00
Sebastian Dröge
30a5987c9e rtp: mp4gpay: Don't set seqnum-base on the caps 
This is supposed to be set by another layer, e.g. rtspsrc.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1693>
 2024-08-10 08:06:40 +00:00
Sebastian Dröge
de42ae432c rtp: basepay: Fix off-by-one with seqnum-offset 
Setting a seqnum-offset of 1 would've caused the first packet to have a
seqnum of 2 instead of 1.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1693>
 2024-08-10 08:06:40 +00:00
Sebastian Dröge
c5163a73ee rtp: basepay: Don't negotiate twice in the beginning 
If srcpad caps are already set as part of sinkpad caps handling, unset
the reconfigure flag so negotiation does not happen yet another time on
the first buffer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1693>
 2024-08-10 08:06:40 +00:00
Sebastian Dröge
31e836f4d6 rtp: basepay: Negotiate SSRC and PT with downstream if not set via property 
This makes the new payloaders closer to the old ones, and makes usage in
webrtcbin easier.

Also properly configure default PT of subclasses. Previously any PT that
was set for these subclasses via g_object_new() would be overridden by
the default one during construction.

Additionally, do SSRC collision handling while queueing output packets.
This is the more natural place as that's where the SSRC is actually
used, it happens potentially earlier and also allows to drain any
pending packets before the SSRC change in the caps.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/557

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1693>
 2024-08-10 08:06:40 +00:00
Sebastian Dröge
914ffc8be9 rtp: basepay: Initialize class fields 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1693>
 2024-08-10 08:06:40 +00:00
Sebastian Dröge
c554a5dc76 rtp: basepay: Don't unset stats on FlushStop 
They are still valid and unsetting them here would cause no stats to
ever be updated again until the next state change.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1693>
 2024-08-10 08:06:40 +00:00
Sebastian Dröge
035a199109 rtp: basepay: Don't use suggested SSRC on collissions if it's the current one 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1693>
 2024-08-10 08:06:40 +00:00
Sebastian Dröge
fa060b9fa0 Fix various 1.80 clippy warnings 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1688>
 2024-08-05 14:14:17 +03:00
Sebastian Dröge
d4d02d70a8 rtp: Require bitstream-io < 2.4.0 
Version 2.4.0 contains a breaking change that it shouldn't, and updating
to 2.4.0 requires a newer Rust version.

See https://github.com/tuffy/bitstream-io/issues/22
 2024-07-16 19:13:49 +03:00
Sebastian Dröge
98b28d69ce Update for new debug log macro syntax 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1658>
 2024-07-08 11:25:23 +03:00
Philippe Normand
eee93aea52 rtp2: Fix typo on auto-header-extension property name 
The rtp (de)pay elements use auto-header-extension so the new elements should do
the same.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1646>
 2024-07-02 09:35:39 +01:00
Matthew Waters
39b61195ad rtprecv: ensure that stopping the rtp src task does not critical 
When pad a released, then we were removing the pad from an internal
list. If the pad was not already deactivated, the deactiviation would
attempt to look for the pad in that list and panic if it was not there.

Fix by delaying removal of the pad from the list until after pad
deactivation occurs.

Also includes test.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1618>
 2024-06-24 13:13:28 +00:00
Matthew Waters
10a31a397e rtp/recv: support pushing buffer lists from the jitterbuffer 
Multiple concurrent buffers produced by the jitterbuffer will be
combined into a single buffer list which will be sent downstream.

Events or queries that interrupt the buffer flow will cause a split in
the output buffer list.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1618>
 2024-06-24 13:13:28 +00:00
Matthew Waters
d036abb7d2 rtp/recv: support buffers lists on rtp sink pad 
In one case, improves throughput by 25% when buffer lists are used.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1618>
 2024-06-24 13:13:28 +00:00
Matthew Waters
df4a4fb2ef rtp/send: support receiving buffer lists 
Can reduce processing overhead if many buffers are pushed concurrently.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1618>
 2024-06-24 13:13:28 +00:00
Matthew Waters
2d1f556794 rtp/session: guard against a busy wait with no members 
If the number of members is 0, then the calculated time to the next rtcp
wakup would be 'now' and could result in a busy loop in the rtcp
processing.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1618>
 2024-06-24 13:13:28 +00:00
Matthew Waters
84a9f9c61f rtp/source: use extended sequence number helper 
Instead of rolling our own

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1618>
 2024-06-24 13:13:28 +00:00
Sebastian Dröge
9b323a6519 Use Option::is_some_and(...) instead of Option::map_or(false, ...) 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1630>
 2024-06-19 13:03:37 +00:00
Sebastian Dröge
4677948a82 rtp: av1pay: Derive Default trait for the state instead of manual implementation 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1624>
 2024-06-18 08:07:24 +03:00
Sebastian Dröge
d357a63bf9 rtp: av1pay: Correctly use N flag for marking keyframes 
The "first packet of a coded video sequence" means that this should be
the first packet of a keyframe that comes together with a sequence
header, not the first packet of a new frame.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/558

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1624>
 2024-06-18 08:06:59 +03:00
Sebastian Dröge
5cd9e34265 rtp: av1pay: Correctly skip over ignored OBUs 
The reader is already after the header at this point so only the OBU
content has to be skipped.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1624>
 2024-06-18 08:06:59 +03:00
Sebastian Dröge
bbe38b9599 rtp: av1: Drop padding OBUs too like Chrome does 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1624>
 2024-06-18 08:06:59 +03:00
Sebastian Dröge
343680ffea rtp: av1depay: Don't return an error if parsing a packet fails 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1612>
 2024-06-14 13:13:21 +00:00
Sebastian Dröge
477855789d rtp: av1depay: Also log warnings on errors 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1612>
 2024-06-14 13:13:21 +00:00
Sebastian Dröge
93c9821cba rtp: av1depay: Drop unusable packets as early as possible 
Otherwise they would pile up until a discontinuity or until we can
actually output something.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1612>
 2024-06-14 13:13:21 +00:00
Sebastian Dröge
0ca4a3778a rtp: av1depay: Parse internal size fields of OBUs and handle them 
They're not recommended by the spec to include in the RTP packets but it
is valid to include them. Pion is including them.

When parsing the size fields also make sure to only take that much of a
payload unit and to skip any trailing data (which should not exist in
the first place).

Pion is also currently storing multiple OBUs in a single payload unit,
which is not allowed by the spec but can be easily handled with this
code now.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/560

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1612>
 2024-06-14 13:13:21 +00:00
Sebastian Dröge
69c3c2ae46 Fix various new clippy 1.79 warnings 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1620>
 2024-06-14 08:33:49 +03:00
Matthew Waters
260b04a1cf rtpbin2: protoct against adding with overflow 
If jitter is really bad, then this calculation may overflow.  Protect
against that.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1605>
 2024-06-06 11:43:26 +00:00
Tim-Philipp Müller
ab2f5e3d8d rtp: ac3: add some unit tests 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1586>
 2024-06-01 12:43:27 +00:00