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Author SHA1 Message Date
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
Mathieu Duponchelle fffd7dc542 webrtc/README: update command to run the signalling server 
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/277

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1017>
 2022-12-16 18:51:08 +02:00
Sebastian Dröge b4185134d1 Fix various new clippy warnings 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1017>
 2022-12-16 18:51:00 +02:00
Sebastian Dröge 7b1ee9f948 webrtchttp: Remove unnecessary clippy warning override 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1008>
 2022-12-12 14:31:33 +02:00
Sebastian Dröge 8c27aefe76 net: Update to async-tungstenite 0.19 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1008>
 2022-12-12 13:39:38 +02:00
Sebastian Dröge 412c191fc2 whipsink: Handle offer creation errors more gracefully 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1008>
 2022-12-12 13:39:16 +02:00
Sebastian Dröge e46d2dfa54 webrtchttp: Fix missing import for docs build 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1008>
 2022-12-12 13:39:10 +02:00
Sebastian Dröge e4788662b9 webrtchttp: Don't use let-else for now 
We still support Rust 1.63.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1008>
 2022-12-12 13:39:04 +02:00
Sebastian Dröge cab5410782 webrtchttp: Fix formatting 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1008>
 2022-12-12 13:38:59 +02:00
Sanchayan Maity 8ac5632561 webrtchttp: Use tokio runtime for spawning thread used for candidate offer 
While at it, we had a bug in whepsrc where for redirect we were
incorrectly calling initial_post_request instead of do_post. Fix
that.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1008>
 2022-12-12 13:38:53 +02:00
Sanchayan Maity 4f67623c22 webrtchttp: Use a proper Rust type name for ICE transport policy 
We don't need to namespace here but can just use the Rust namespaces.
Only the GType name has to stay like it is.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1008>
 2022-12-12 13:38:48 +02:00
Sanchayan Maity 1d4d9b3bdb webrtchttp: Do not import element_imp_error 
element_imp_error and such macros should not be imported but rather
only be accessed via gst namespace.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1008>
 2022-12-12 13:38:41 +02:00
Sanchayan Maity 3202c4dc39 webrtchttp: Do not block webrtcbin signal handlers for sending candidates 
While at it, drop the OPTIONS request in WHIP sink. This was not really
required. See section 4.4 of the spec
https://www.ietf.org/archive/id/draft-ietf-wish-whip-01.html#name-stun-turn-server-configurat

Also introduce a new error type and distinguish between a future being
aborted or returning an error.

We call abort only during shutdown and hence except for the DELETE
resource request being aborted, other waits on future should not
be fatal.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1008>
 2022-12-12 13:38:36 +02:00