This commit adds support for raw payloads such as L24 audio to `webrtcsink` &
`webrtcsrc`.
Most changes take place within the `Codec` helper structure:
* A `Codec` can now advertise a depayloader. This also ensures that a format
not only can be decoded when necessary, but it can also be depayloaded in the
first place.
* It is possible to declare raw `Codec`s, meaning that their caps are compatible
with a payloader and a depayloader without the need for an encoder and decoder.
* Previous accessor `has_decoder` was renamed as `can_be_received` to account
for codecs which can be handled by an available depayloader with or without
the need for a decoder.
* New codecs were added for the following formats:
* L24, L16, L8 audio.
* RAW video.
The `webrtc-precise-sync` examples were updated to demonstrate streaming of raw
audio or video.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1501>
The certificate chain was incorrectly being passed the private key instead
of certificate. With rustls 0.23.11 version, this error was being caught
and reported. As stated in the 0.23.11 release, it has a new feature
"API for determining whether a CertifiedKey's certificate and private key
matches: keys_match(). This is called from existing fallible functions
that accept a private key and certificate (for example, with_single_cert())
so these functions now detect this misconfiguration."
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1666>
This element allows wrapping an existing live "mpeg-ts source" (udpsrc,
srtsrc,...) and providing a clock based on the actual PCR of the stream.
Combined with `tsdemux ignore-pcr=True` downstream of it, this allows playing
back the content at the same rate as the (remote) provider **and** not modify
the original timestamps.
Co-authored-by: Sebastian Dröge <slomo@coaxion.net>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1640>
As suggested in the aws crate documentation, wrap SDK errors with
DisplayErrorContext so their Display implementation outputs the full
context.
Improve error display from "dispatch failure" to
"dispatch failure: io error: error trying to connect: dns error: failed
to lookup address information: Name or service not known: dns error:
failed to lookup address information: Name or service not known: failed
to lookup address information: Name or service not known
(DispatchFailure(DispatchFailure { source: ConnectorError { kind: Io,
source: hyper::Error(Connect, ConnectError(\"dns error\", Custom { kind:
Uncategorized, error: \"failed to lookup address information: Name or
service not known\" })), connection: Unknown } }))"
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1638>
We now check if the peer actually supports Datagram and refusing to
proceed if it does not. Since the datagram size can actually change
over the lifetime of a connection according to variation in path MTU
estimate, also check buffer size before trying to send.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1613>
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>
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>
We can use `is_some_and(...)` instead of `map_or(false, ...)`.
Also in a few places the factory was retrieved multiple times, one time
with unwrapping and another time with handling the `None` case
correctly. Instead of unwrapping, move code to handle the `None` case.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1630>
We lost the environment variable checks during the addition of the
putobjectsink tests, which caused failures on MR branches.
It would be nicer to use some other mechanism to validate the tests can
run, so we don't count on only the environmnent, but for now this will
have to do.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1629>
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>
It would be possible that there is no cancellable yet when unlock() is
called, then a new future is executed and it wouldn't have any
information that it is not supposed to run at all.
To solve this remember if cancellation should happen and reset this
later.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1602>
It would be possible that there is no cancellable yet when unlock() is
called, then a new future is executed and it wouldn't have any
information that it is not supposed to run at all.
To solve this remember if unlock() was called and reset this in
unlock_stop().
Also implement actual unlocking in s3hlssink.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1602>
It would be possible that there is no cancellable yet when unlock() is
called, then a new future is executed and it wouldn't have any
information that it is not supposed to run at all.
To solve this remember if unlock() was called and reset this in
unlock_stop().
Also actually implement unlock() / unlock_stop() for the sink, and don't
cancel in stop() as unlock() / unlock_stop() would've been called before
that already.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1602>
It would be possible that there is no cancellable yet when unlock() is
called, then a new future is executed and it wouldn't have any
information that it is not supposed to run at all.
To solve this remember if unlock() was called and reset this in
unlock_stop().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1602>
- generate a new session id for every new client
use the session id in the resource url
- remove the producer-peer-id property in the WhipServer signaler as it
is redundant to have producer id in a session having only one producer
- read the 'producer-peer-id' property on the signaller conditionally
if it exists else use the session id as producer id
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1339>
- Add a new structure Session
- manage each producer using a session
- avoid send EOS when a session terminates, instead keep running
waiting for any new producer to connect
- Maintain a bin element per session
- each session bin encapsulates webrtcbin and the decoder if needed
as well as the parser and filter if requested by the application
(through request-encoded-filter)
- this will be helpful to cleanup the session's respective elements
when the corresponding producer terminates the session
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1339>
There is now two elements, rtpsend and rtprecv that represent the two
halves of a rtpsession. This avoids the potential pipeline loop if two
peers are sending/receiving data towards each other. The two halves can
be connected by setting the rtp-id property on each element to the same
value and they will behave like a combined rtpbin-like element.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
Can receive and recevie one or more RTP sessions containing multiple
pt/ssrc combinations.
Demultiplexing happens internally instead of relying on separate
elements.
Co-Authored-By: François Laignel <francois@centricular.com>
Co-Authored-By: Mathieu Duponchelle <mathieu@centricular.com>
Co-Authored-By: Sebastian Dröge <sebastian@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
Add to awss3sink and awss3putobjectsink elements the following
paramerters which are set on the uploaded S3 objects:
* cache-control;
* content-encoding; and
* content-language
Bugfix: Set the content-type and content-disposition values in the S3
putobject call. Previously the params were defined on the element but
unused.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1585>
Unit tests specify a 0-based offset, so printing that plus the
random initial offset on failure is just needlessly confusing,
so subtract the initial offset when printing expected/actual
values. The real values are still printed as part of the assert.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1580>
