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Author SHA1 Message Date
Mathieu Duponchelle
0954af10c7 webrtc/signalling: fix race condition in message ordering 
Spawning one task per message to send out instead of sending them out
sequentially from the one task used to poll the handler sometimes
resulted in peers receiving ICE candidates before SDP offers, triggering
hard to understand errors in the browser.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1254>
 2023-06-20 22:30:01 +02:00
Sebastian Dröge
dfe2442c92 webrtc/signalling: Allow unknown clippy lints 
tracing is adding some that require a newer Rust version than used here.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1249>
 2023-06-19 20:37:53 +03:00
Mathieu Duponchelle
82f3910453 webrtcsink: don't try to use cudaconvert if not present 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1249>
 2023-06-19 19:03:04 +03:00
Mathieu Duponchelle
55a6609fdb webrtcsrc: add twcc extension to codec-preferences when present 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1249>
 2023-06-19 19:02:48 +03:00
Sebastian Dröge
05b2caec74 webrtc: Update to fastrand 2 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1249>
 2023-06-19 19:02:16 +03:00
Sebastian Dröge
c9b2c88469 Use MPL as license specifier for plugins only requiring GStreamer < 1.20 
And use MPL-2.0 for all that require GStreamer 1.20 or newer. The new
string is only allowed in 1.20 or newer and using it in older versions
causes failure to load the plugin.

All affected plugins are of course still MPL-2.0 licensed.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/374

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1249>
 2023-06-19 19:01:52 +03:00
Mathieu Duponchelle
e64e12e478 net/aws/transcriber: track discont offset in input stream 
and add it up to subsequent transcripts.

This ensures synchronization is maintained even after the input stream
experiences a discontinuity and a gap in its timestamps.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1234>
 2023-06-06 23:45:05 +03:00
Edward Hervey
60ae3fc0b9 rtpgccbwe: Improve packet handling 
Both the delay-based *and* loss-based estimates should be computed instead of
just one. This ensures faster adaptation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1234>
 2023-06-06 22:53:07 +03:00
Sebastian Dröge
2002c54582 whipsink: Request pads with webrtcbin's pad templates and not our own 
It's invalid to request pads with a pad template that is not part of the
element, and results in a critical warning.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1234>
 2023-06-06 22:52:53 +03:00
Mathieu Duponchelle
8248425905 webrtcsink: further refactor connection to stats signals 
- Stop passing webrtcbin around without using it

- Stop using glib::closure as clippy complains when using a unit type
  default-return

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1234>
 2023-06-06 22:52:43 +03:00
Mathieu Duponchelle
e9d32fb221 webrtcsink: fix stats_sigid logic 
First off, we just created the session, we know stats_sigid is None
at this point.

Second, don't first assign the result of connecting on-new-ssrc to the
field, then the result of connection twcc-stats, that simply doesn't
make sense.

Finally, actually check that stats_sigid *is* None before connecting
twcc-stats, as I understand it this must have been the original
intention / behavior.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1234>
 2023-06-06 22:52:34 +03:00
Mathieu Duponchelle
8ff2c6609c webrtcsink: don't panic in twcc-stats callback 
If webrtcbin was disposed of at this point, simply return

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/345
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1234>
 2023-06-06 22:52:22 +03:00
Thibault Saunier
0b65a2f8af webrtcsrc: Do not pass raw caps in the transceiver 
That was not making sense.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1216>
 2023-05-18 18:30:08 +03:00
Thibault Saunier
482ff879a4 webrtcsrc: Fix caps used when creating transceiver 
We used to pass all media keys and attributes to the caps which
incorrect. Instead we should be using only the keys from the map
and remove all information related to rtcp which is irrelevant
to create the transceiver.

This also simplifies the code.

New caps look like:

```
Caps(
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 96,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "VP8",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 102,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "H264",
        packetization-mode: (gchararray) "1",
        profile: (gchararray) "baseline",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 104,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "H264",
        packetization-mode: (gchararray) "0",
        profile: (gchararray) "baseline",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 106,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "H264",
        packetization-mode: (gchararray) "1",
        profile: (gchararray) "constrained-baseline",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 108,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "H264",
        packetization-mode: (gchararray) "0",
        profile: (gchararray) "constrained-baseline",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 127,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "H264",
        packetization-mode: (gchararray) "1",
        profile: (gchararray) "main",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 39,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "H264",
        packetization-mode: (gchararray) "0",
        profile: (gchararray) "main",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 98,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "VP9",
        profile-id: (gchararray) "0",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 100,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "VP9",
        profile-id: (gchararray) "2",
    },
)
```

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1216>
 2023-05-18 18:30:08 +03:00
Edward Hervey
f4a565d4ea rtpgccbwe: Don't process empty lists 
The structure parsing could result in an empty vector. Don't do any processing
since the loss code assumes it's non-empty for average estimates which would
result in weird/invalid results.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1190>
 2023-04-22 12:10:43 +03:00
Sebastian Dröge
e0a7c93d46 ndisrc: Fix copying of raw video frames with different NDI/GStreamer strides 
And also don't copy each line twice for single-plane formats.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1161>
 2023-04-05 18:17:31 +03:00
Tim-Philipp Müller
2d56989f5c git: replace LICENSE file symlinks with copies 
Git will de-duplicate the contents for us anyway, and
symlinks can cause problems with some versions of git
and also on Windows.

https://github.com/mesonbuild/meson/issues/11646
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4326

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1161>
 2023-04-05 18:17:16 +03:00
Mathieu Duponchelle
c2d6273786 webrtcsink: fix calculation of fec_ratio with multiple encoders 
In this context, the bitrate variable is for all encoders, but the
max_bitrate field is per encoder. To calculate a proper FEC ratio, we
need to scale max_bitrate to the number of encoders.

+ Also clamp the fec-percentage that we set on the transceiver for extra
  safety

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1161>
 2023-04-05 18:16:10 +03:00
Thibault Saunier
4b867d27fe Add a webrtcsrc element 
Updating the docker image to include:
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3236

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1117>
 2023-03-02 14:56:30 -03:00
Sebastian Dröge
9a779607c7 Update versions to 0.9.10 2023-03-02 13:18:00 +02:00
Vivia Nikolaidou
a0fe1aba5f ndisinkcombiner: Properly handle caps changes 
We are caching one video buffer, so previously we were changing the src
caps one buffer too early.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1116>
 2023-03-02 11:01:18 +02:00
Thibault Saunier
e4c9ba43df webrtc: Enhance debug messages when using unknown peer ID 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1116>
 2023-03-02 11:01:18 +02:00
Matthew Waters
0d3dc25414 webrtcsink: also support nvvidconv in lieu of nvvideoconvert 
nvvideoconvert may not exist and nvvidconv might on some Jetson
platforms.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1116>
 2023-03-02 10:53:19 +02:00
Arun Raghavan
611c7d6cd3 hlssink3: Allow GIOStream signal handlers to return None 
If creating a playlist or fragment stream fails (disk is full, the
directory is removed, ...), we will currently crash because the signal
handler expects a non-None GIOStream. The actual callback is allowed to
return None values and we handle this in the caller, so let's not have
this restriction on the signal handler.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1098>
 2023-02-21 16:17:45 +02:00
Seungha Yang
562b429388 rtpav1pay: Fix Leb128Bytes size parsing 
There are multiple ways of encoding the value, and don't assume
that bitstream used the way used in this plugin. Instead, count
the number of used bytes.

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/312
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1092>
 2023-02-11 19:44:51 +02:00
Sebastian Dröge
eb3d3b3088 Update versions to 0.9.9 2023-02-09 22:08:17 +02:00
rajneeshksoni
01d3b0f9da awss3sink: Add properties to set content-Type and content-disposition. 
for uploaded object default content-type is set to binary/octet-stream,
which is correct.
metadata cannot be used to set content-type and content-disposition as
setting metadata add a prefix x-amz-meta to key
e.g. setting metadate "content-type=video/mp4" actually set value as
x-amz-meta-content-type. So these has to be seaprate property.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1086>
 2023-02-09 21:43:57 +02:00
rajneeshksoni
f96b64e1c1 hlssink3: Allow setting i-frame-only playlist. 
HLS allows manifest where all segments are single ifames.
This manifest requires `EXT-X-I-FRAMES-ONLY` tag in the
manifest.
I-FRAMES-ONLY playlist segments are video only segments.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1086>
 2023-02-09 21:43:57 +02:00
Sebastian Dröge
5f70c0f5fe rtpgccbwe: Don't use clamp() if there's no clear min/max value 
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/305

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1086>
 2023-02-09 21:43:57 +02:00
Sanchayan Maity
0e55e19d57 aws/s3hlssink: Fix deadlock on EOS 
In state change to NULL, we take state lock and call stop. When stop
is called, we will try to upload queued segments in S3 request thread.
That tries to take the state lock again and deadlocks.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1086>
 2023-02-09 21:14:06 +02:00
Sanchayan Maity
7a8ecb5343 aws/s3hlssink: Use factory name when checking name of child element 
Commit ad3f1cf fixed the name of hlssink child element to be the same
for hlssink2 and hlssink3. However, we rely on element name to return
boolean in case of hlssink3 or None in case of hlssink2 as the return
value of the delete-fragment closure.

Fix this by using the factory name instead of the element name.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1086>
 2023-02-09 21:14:01 +02:00
Sebastian Dröge
17dec1cb26 rtpav1pay: Add support for tu/frame aligned input 
In this case every buffer can be sent out immediately and makes up a
whole frame.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1086>
 2023-02-09 21:13:40 +02:00
Sebastian Dröge
ba0904630d rtpav1pay: Consider the marker flag to output packets immediately at the end of a frame 
Otherwise it is necessary to wait for the beginning of the following
frame, which unnecessarily increases the latency.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/255

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1086>
 2023-02-09 21:13:33 +02:00
Sebastian Dröge
af0e6281d2 rtpav1depay: Fix depayloading of packets starting with a leading OBU fragment followed by more OBUs 
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/288

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1086>
 2023-02-09 21:13:26 +02:00
Sebastian Dröge
e79221f386 rtpav1depay: Fix error handling 
Don't error out immediately on errors anymore but try again with the
next packet.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/289

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1086>
 2023-02-09 21:13:19 +02:00
Sebastian Dröge
dc47b35536 rtpav1depay: Set DISCONT flag on buffers following a corrupted packet 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1086>
 2023-02-09 21:13:13 +02:00
Sebastian Dröge
3520fc67de rtpav1depay: Don't output full TUs but just OBUs as they come 
Simplifies state tracking and potentially reduces latency as it's not
necessary to wait until all fragments of an OBU are received.

The last OBU of a TU is marked with the marker flag to allow parsers to
detect this without first seeing the beginning of the next TU.

Also use a simple `Vec` for collecting complete OBUs instead of a
`gst_base::Adapter` as this reduces the number of allocations.

And also handle invalid packets a little bit more gracefully.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/244

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1086>
 2023-02-09 21:13:06 +02:00
Arun Raghavan
f3b8288ef9 aws: s3hlssink: Fix the name of the hlssink child element 
It's easier to set child element properties if the name doesn't depend
on the factory.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1086>
 2023-02-09 21:05:58 +02:00
Sebastian Dröge
5c2582d105 Update version to 0.9.8 2023-01-23 11:30:27 +02:00
Sebastian Dröge
4ba452dcc3 Update versions to 0.9.7 2023-01-19 19:06:43 +02:00
Sebastian Dröge
c818a575b4 Update versions to 0.9.6 2023-01-18 17:19:17 +02:00
Sebastian Dröge
d02508a7d0 aws: Update to AWS SDK 0.53/0.23 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1050>
 2023-01-18 16:56:10 +02:00
Mathieu Duponchelle
53ae335d22 webrtcsink: fix panic on pre-bwe request error 
We dispose of consumer pipelines asynchronously, potentially after the
session objects have been disposed of.

As session objects are the owner of the cc element, it is entirely
possible for the bwe-request signal to get emitted after cc has been
disposed of, as the closure only takes a weak reference to it.

Fix by simply checking if cc is None

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1045>
 2023-01-11 18:38:13 +02:00
Sebastian Dröge
2a8a90f76f Update versions to 0.9.5 2023-01-07 16:06:17 +02:00
Sebastian Dröge
4b936950c2 aws: Update to test-with 0.9 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1038>
 2023-01-07 13:25:37 +02:00
rajneeshksoni
698ab100b3 awss3hlssink: Add stats property. 
application can monitor the progress of hls segment generation
and upload progress.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1038>
 2023-01-07 13:25:31 +02:00
Philippe Normand
517dc286d0 rtpav1depay: Implement srcpad set_caps 
Without this auto-pluggers such as decodebin or parsebin will be unable to
process AV1 RTP payloads.

Tested with: `videotestsrc num-buffers=50 ! videoconvert ! av1enc ! av1parse ! rtpav1pay ! queue ! decodebin3 ! videoconvert ! queue ! autovideosink`

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1038>
 2023-01-07 13:25:15 +02:00
Sebastian Dröge
b0bd55c4d2 Update versions to 0.9.4 2022-12-27 13:14:59 +02:00
Sebastian Dröge
deeff67f94 aws: Update to AWS SDK 0.52/0.22 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1027>
 2022-12-27 12:39:56 +02:00
Sebastian Dröge
bae5294e8f Update versions to 0.9.3 2022-12-16 20:22:17 +02:00