Commit graph

3496 commits

Author SHA1 Message Date
Sebastian Dröge
fc29ff7d8b hlssink3: Update to sprintf 0.2
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1757>
2024-08-27 21:06:52 +03:00
Sebastian Dröge
52895223a8 fmp4: Update to dash-mpd 0.17
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1757>
2024-08-27 21:02:43 +03:00
Sebastian Dröge
7e9a6d6893 Update Cargo.lock
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1757>
2024-08-27 20:58:48 +03:00
Mathieu Duponchelle
2f9bb62b6b gstwebrtc-api: create control data channel when offering
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1755>
2024-08-27 07:52:12 +02:00
Sanchayan Maity
f0d42b88cb Add s3putobjectsink to README
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1550>
2024-08-26 19:56:34 +00:00
Sanchayan Maity
f3206c2e1a aws: Add next-file support to putobjectsink
Add `next-file` support to `awss3putobjectsink` on similar lines to
the `next-file` support in `multifilesink`.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1550>
2024-08-26 19:56:34 +00:00
Sanchayan Maity
d274caeb35 whepsrc: Fix incorrect default caps
add-transceiver needs application/x-rtp caps and not raw caps. We were
providing raw caps which is incorrect.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1748>
2024-08-26 19:44:37 +05:30
Mathieu Duponchelle
66727188cf net/aws: fix sanity check in transcribe loop
When we receive a new alternative we want to avoid iterating out of
bounds, but the comparison between the current index and the length of
the alternative should not log an error when partial_index == length, as
Vec::drain(length..) is valid, and it is completely valid for AWS to
send us a new alternative with as many items as we have already
dequeued.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1751>
2024-08-26 11:37:08 +02:00
Sebastian Dröge
7e912b0dde video/gtk4: Clean up imports a bit
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1708>
2024-08-23 08:11:15 +00:00
Sebastian Dröge
3fd3a32b6f video/gtk4: Clean up GStreamer GL context initialization
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1708>
2024-08-23 08:11:15 +00:00
Jordan Petridis
4f01421cc4 video/gtk4: Make feature "v4_6" the default assumption
It's our minimum gtk4 version we target anyway so we
can remove some of the extra features in the toml and require
it by default.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1708>
2024-08-23 08:11:15 +00:00
Jordan Petridis
ff5032934d video/gtk4: Remove some indentation in frame.rs
We have less cfg features now and don't need the extra
brackets to guard the codepaths.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1708>
2024-08-23 08:11:15 +00:00
Jordan Petridis
602760d0d8 video/gtk4: Remove cfg flag for gst-gl
GTK 4 hard depends on GL on all platforms, and now both
windows and macos have codepaths for the paintable sink to
produce GLTextures.

This we can now drop the cfg build flag we have making it optional.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1708>
2024-08-23 08:11:15 +00:00
Sanchayan Maity
320f36a462 hlssink3: Use fragment duration from splitmuxsink if available
splitmuxsink now reports fragment offset and duration in the
splitmuxsink-fragment-closed message. Use this duration value
for the MediaSegment when available.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1728>
2024-08-22 15:13:21 +00:00
Mathieu Duponchelle
4cf93ccbdb net/webrtc: Add missing npm command to README
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/589

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1746>
2024-08-22 15:46:28 +02:00
Sebastian Dröge
87b72f768b livesync: Report latency query results when handling latency query too
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1739>
2024-08-22 07:33:47 +00:00
Jerome Colle
dee0e32dde webrtcsink: add nvv4l2av1enc support
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1735>
2024-08-22 06:41:52 +00:00
Mathieu Duponchelle
8ad882bed5 gstwebrtc-api: address issues raised by mix matrix support
1c48d7065d was mistakenly merged too
early, and there were concerns about the implementation and API design:

The fact that the frontend had to expose a text area specifically for
sending over a mix matrix, and had to manually edit in floats into the
stringified JSON was suboptimal.

Said text area was always present even when remote control was not
enabled.

The sendControlRequest API was made more complex than needed by
accepting an optional stringifier callback.

This patch addresses all those concerns:

The deserialization code in webrtcsink is now made more clever and
robust by first having it pick a numerical type to coerce to when
deserializing arrays with numbers, then making sure it doesn't allow
mixed types in arrays (or arrays of arrays as those too must share
the same inner value type).

The frontend side simply sends over strings wrapped with a request
message envelope to the backend.

The request text area is only shown when remote control is enabled.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1725>
2024-08-22 05:54:46 +00:00
Piotr Brzeziński
c4bcdea830 hlscmafsink: Add new-playlist signal
Allows you to switch output between folders without having to state change to READY to close the current playlist.
Closes the current playlist immediately and starts a new one at the currently set location.
Should be used after changing the relevant location properties.
Makes use of the send-headers signal in cmafmux.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1692>
2024-08-22 02:06:51 +00:00
Piotr Brzeziński
798936afc9 cmafmux: Add send-headers signal
Forces cmafmux to output headers for the init segment again, alongside the next chunk.
Needed for hlscmafsink to support changing output paths on the fly, without going back to READY.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1692>
2024-08-22 02:06:51 +00:00
Piotr Brzeziński
ad0a23fee7 cmafmux: Add opus support
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1692>
2024-08-22 02:06:51 +00:00
Jordan Petridis
c43fc2d576 ci: Pin the windows jobs to gstreamer runners for now
There seem to be some seemingly random failures if the
jobs get scheduled on the other windows runners,
and they need to be debugged further.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1736>
2024-08-22 01:17:18 +00:00
Jordan Petridis
b9fcb99cd4 ci: Update the .cargo/config file
```
warning: `/builds/alatiera/gst-plugins-rs/.cargo/config` is deprecated in favor of `config.toml`
```

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1736>
2024-08-22 01:17:18 +00:00
Jordan Petridis
b4f22a52ff ci: Add a default retry policy for jobs
Automatically retry if it's a system failure or similar

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1736>
2024-08-22 01:17:18 +00:00
Mathieu Duponchelle
5dc2d56c0e webrtcsink: store mids per-session instead of globally
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1730>
2024-08-21 21:20:40 +00:00
Mathieu Duponchelle
16ee51621e webrtcsink: fix segment format mismatch with remote offer
webrtcsink was starting the negotiation process on Ready and concurrently
moving the consumer pipeline to Playing, but when answering the remote
description was set so fast that input streams were connected (and the time
format set on appsrc) before the state change to Paused had completed.

This meant gst_base_src_start was happening after that and setting the format
back to bytes, the time segment that was next coming in then caused:

basesrc gstbasesrc.c:4255:gst_base_src_push_segment:<video_0> segment format mismatched, ignore

And the consumer pipeline errored out.

The same issue existed in theory when webrtcsink was creating the offer,
but was much harder to trigger as it required that the remote answer
came in before the state change to Paused had completed.

This commit fixes the issue by simply waiting for the state to have
changed to Paused before negotiating.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1730>
2024-08-21 21:20:40 +00:00
Piotr Brzeziński
b6406013c5 hlssink3: Fix racy test by separating events (signals) from bus messages
Was regularly failing on the CI. Bus messages are handled async here, so they need to be tracked separately.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1737>
2024-08-21 19:49:09 +00:00
Mathieu Duponchelle
170e769812 audio: add speechmatics transcriber
Element implemented around the Speechmatics API:

<https://docs.speechmatics.com/rt-api-ref>

The element also comes with translation support, and offers a similar
interface to the one exposed by `awstranscriber`.

The Speechmatics service has good accuracy, and can be deployed on
premises, offering an advantage over AWS transcribe.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1665>
2024-08-21 17:43:02 +00:00
Jordan Petridis
4f69dcd210 ci: Remove leftover scripts
Both of these have been moved in the main image for a while now

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1733>
2024-08-21 06:44:52 +00:00
Piotr Brzeziński
982a9a9aea hlssink3: Post hls-segment-added message
Posts a simple 'hls-segment-added' message with the segment location, start running time and duration.
With hlssink2, it was possible to catch 'splitmuxsink-fragment-closed', but since hlssink3 doesn't forward that message
(and hlscmafsink doesn't even use that mux), the new one was added to allow for listening for new fragments being added.

I extended the existing tests to check whether this message is posted correctly.
They theoretically only cover hlssink3, but hlscmafsink uses the same base class so it should be alright for now.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1677>
2024-08-20 18:32:59 +00:00
Jordan Petridis
5172e8e520 ci: Use the windows specific image tags
Followup to c5dfc87953

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1731>
2024-08-20 17:21:20 +03:00
Sebastian Dröge
eb0a44fe67 ndisrc: Move timestamp handling from demuxer to source
This allows putting correct timestamps on buffers coming out of the
source already instead of leaving them unset until the demuxer.

And also calculate timestamps for metadata buffers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1718>
2024-08-16 06:07:35 +00:00
Mathieu Duponchelle
1c48d7065d gstwebrtc-api example: add support for requesting mix matrix
This is one example of how a consumer might send over custom upstream
event requests to the producer.

As webrtcsink will deserialize numbers in priority as integers, we need
a custom stringifying function to ensure members of the matrix array are
indeed serialized with the floating point.

An optional stringifier parameter is thus added to the
sendControlRequest API.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1711>
2024-08-15 15:42:04 +00:00
Mathieu Duponchelle
01e28ddfe2 webrtcsink: implement generic data channel control mechanism ..
.. and deprecate data channel navigation in favor of it.

A new property, "enable-data-channel-control" is exposed, when set to
TRUE a control data channel is offered, over which can be sent typed
upstream events.

This means further upstream events will be usable, for now only
navigation and custom upstream events are handled.

In addition, send response messages to notify the consumer of whether
its requests have been handled.

In the future this can also be extended to allow the consumer to send
queries, or seek events ..

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1711>
2024-08-15 15:42:04 +00:00
Tim-Philipp Müller
0a4dc29efe ci: tag cerbero trigger job as placeholder job 2024-08-14 17:23:59 +01:00
Jordan Petridis
086281b03d ci: Update ci-template sha
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1721>
2024-08-14 18:23:48 +03:00
Mathieu Duponchelle
0a6963f7ce gstwebrtc-api: example: use http by default
That way the webpage connects with ws:/ to the signaller.

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/589
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1704>
2024-08-14 14:10:04 +00:00
Sebastian Dröge
102185d09d mpegtslivesrc: Handle PCR discontinuities as errors for now
More work is needed to make this work seemlessly and right now it would
simply cause invalid timestamps to be created.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1717>
2024-08-14 12:34:18 +00:00
Sebastian Dröge
ede82ca5b4 hlssink3: Don't use is-live=true
This sometimes produces imperfect timestamps that cause the fragment
duration to be slightly different than expected.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1716>
2024-08-14 13:05:40 +03:00
Tim-Philipp Müller
e21f341a03 ci: set cerbero trigger job timeout to 4h
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1716>
2024-08-13 20:34:17 +01:00
Guillaume Desmottes
72e53b9f16 videofx: update image and image_hasher deps
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1707>
2024-08-13 07:21:59 +00:00
Guillaume Desmottes
ea29052c39 cdg: update to image 0.25
I just published a new cdg_renderer release depending of image 0.25.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1707>
2024-08-13 07:21:59 +00:00
Jordan Petridis
3e97fef6ce ci: Generate html and cobertura coverage with a single command
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1709>
2024-08-13 06:41:17 +00:00
Sebastian Dröge
bc930122ba webrtcsrc: Make sure to always call end_session() without the state lock
This was already done in another place for the same reason: preventing a
deadlock. It's probably not correct as hinted by the FIXME comment but
better than deadlocking at least.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1701>
2024-08-13 06:04:09 +00:00
Mathieu Duponchelle
0da1c8e9c9 webrtcsink: fix assertions when finalizing
Dumping the pipeline on state changes from an async bus handler was
triggering criticals.

Instead, dump from the sync handler.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1706>
2024-08-12 09:13:06 +02:00
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