Commit graph

2507 commits

Author SHA1 Message Date
Guillaume Desmottes 93503c0ca9 spotify: move Settings to common module
Will be used to implement the lyrics source element.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1095>
2023-03-05 14:35:04 +01:00
Sebastian Dröge cd177dee86 Update CHANGELOG.md for 0.10.3 2023-03-02 13:35:47 +02:00
Vivia Nikolaidou cd74d01324 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/1110>
2023-03-01 12:30:54 +00:00
Sebastian Dröge fb528941ea deny: Update to allow socket2 0.4
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1113>
2023-03-01 14:00:26 +02:00
Sebastian Dröge e69f1e0b8c threadshare: Update to socket2 0.5
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1113>
2023-03-01 13:59:53 +02:00
François Laignel 4a988aaeb8 net/aws/transcriber: use a TranscriberLoop struct
This helps gather together the details related to the `TranscriberLoop`.
One difference with previous implementation is that the ws `Client` is
build each time the loop is started instead of being reused. With the new
approach, we don't keep the connection open after EOS and we should be
more resistant in case of a connection failure.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1104>
2023-03-01 08:47:58 +00:00
François Laignel f1a080c94e net/aws/transcriber: own transcription items
So that we can avoid copying the content.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1104>
2023-03-01 08:47:58 +00:00
François Laignel 36ae29d746 net/aws: enqueue transcribed buffers within the ws loop
Instead of sending transcription events to the src pad loop, this commit
enqueues the transcribed buffers immediately in the ws loop, then notifies
the src pad loop. The src pad loop is only in charge of dequeuing the buffers.

This should help with upcoming evolutions.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1104>
2023-03-01 08:47:58 +00:00
François Laignel 00153754bb net/aws: use aws-sdk-transcribestreaming
Switch from manual webservice client impl to `aws-sdk-transcribestreaming`.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1104>
2023-03-01 08:47:58 +00:00
François Laignel 57f365979c net/aws: remove aws_ from the aws_transcribe* folder names
Those folders reside under `aws`, so there's shouldn't be any confusion.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1104>
2023-03-01 08:47:58 +00:00
Thibault Saunier ce3bb2f1d4 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/932>
2023-02-28 20:50:15 -03:00
Thibault Saunier 0ae637f531 webrtcsink: Move RUNTIME to the crate so it can be reused
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/932>
2023-02-28 17:57:14 -03:00
Thibault Saunier 4ec441560b webrtc: Enhance debug messages when using unknown peer ID
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/932>
2023-02-28 19:28:51 +00:00
Guillaume Desmottes a0f6e84ec2 tracers: queue_levels: add appsrc support
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1111>
2023-02-28 14:38:29 +01:00
Sebastian Dröge ff2f7a8505 livesync: Correctly calculate fallback buffer duration from framerate
Numerator and denominator were switched.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1108>
2023-02-28 12:52:11 +02:00
Matthew Waters 542c7e12b8 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/1107>
2023-02-28 10:12:36 +11:00
Sebastian Dröge bca4af0c79 gtk4: Set sync point on the video frame after mapping it
Otherwise it is not always ready for use yet in GTK even after waiting
on the sync point, and a fully transparent texture is rendered instead.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/320

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1103>
2023-02-24 11:52:28 +02:00
Sebastian Dröge 6bc72e513c Update CHANGELOG.md for 0.10.2 2023-02-23 10:16:55 +02:00
Jordan Petridis 90455a8111 video/gtk4: Add a flatpak snippet example in the README
Close #155

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1102>
2023-02-22 22:15:40 +02:00
Sebastian Dröge ce1faa6020 gtk4: Attach channel receiver to the default main context from the main thread
It requires acquiring the main context for thread-safety reasons and
that is only possible from the main thread itself.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/319

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1099>
2023-02-22 10:02:45 +02:00
Sebastian Dröge f08b65ece1 gtk4: Don't unnecessarily set the sink to READY to retrieve the context
That's not needed and will cause the GL context messages to be not
distributed inside the pipeline.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1099>
2023-02-21 21:36:45 +02:00
Sebastian Dröge 8aa5125d5b gtk4: Refactor and simplify GL context handling
Create a single, global GDK GL context and the corresponding GStreamer
GL display and wrapped GStreamer GL context when initializing the first
sink and continue using that for all further sinks.

Additionally, don't create a full GStreamer GL context inside the sink
but only distribute the wrapped GL context in the pipeline so that
elements that actually need a full GL context can create one that is
sharing with that one. The sink itself does not need a full GStreamer GL
context.

Then inside the sink check that any GL memory that arrives was created
by a GL context that can share with the wrapped GDK GL context and only
then use it.

And lastly, use the correct GL contexts for a) creating a sync point and
b) actually waiting on it.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/318

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1099>
2023-02-21 21:36:45 +02:00
Sebastian Dröge 93051dba0e ci: Update image version
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1096>
2023-02-20 13:08:51 +02:00
Sebastian Dröge 143bf7608e ci: Update to cargo-c 0.9.15
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1096>
2023-02-20 13:08:46 +02:00
Sebastian Dröge 53b6efaa6e ci: Update to dav1d 1.1
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1096>
2023-02-20 11:09:21 +02:00
Sebastian Dröge 9fc1404415 Update minimum supported Rust version to 1.66
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1096>
2023-02-20 11:09:01 +02:00
Seungha Yang 59222f7a35 mp4mux: Ignore framerate update
like mp4mux in -good does already

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1094>
2023-02-16 03:43:01 +09:00
Seungha Yang 6b15e772ac fmp4mux: Ignore framerate update
like mp4mux in -good does already

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1094>
2023-02-16 02:23:56 +09:00
Arun Raghavan 487d7fb26b 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/1093>
2023-02-14 11:25:44 -05:00
Guillaume Desmottes 77e99e92fb spotifyaudiosrc: use Settings Default to define default props
Makes it easier to change one property's default value.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1074>
2023-02-13 18:04:53 +01:00
Sebastian Dröge 51d61af863 Add mp4 plugin to README.md 2023-02-13 11:56:33 +02:00
Sebastian Dröge b0664b3c85 Update CHANGELOG.md for 0.10.1 2023-02-13 11:54:46 +02:00
Sebastian Dröge 04e101c605 Optimize various error message / debug message formatting
Directly make use of format strings instead of formatting a string
beforehand and then passing it to the macros.
2023-02-13 11:50:57 +02:00
Sebastian Dröge 034c0f0fd8 Add CHANGELOG.md for 0.10.0 release
This is the first one and only lists changes from 0.9.0 to 0.10.0
2023-02-12 13:15:17 +02:00
Arun Raghavan 39e0acb55a hlssink3: Fix case on unspecified playlist type nick for consistency
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1089>
2023-02-10 23:07:12 +00:00
Seungha Yang 6420fe43da 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/1090>
2023-02-10 18:47:52 +00:00
Sebastian Dröge ac8afc4ac0 Update to async-tungstenite 0.20
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1087>
2023-02-10 13:03:07 +02:00
Sebastian Dröge 1e13dbb99c Update versions to 0.11.0-alpha.1 2023-02-10 00:23:56 +02:00
rajneeshksoni 994c79569e 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/1085>
2023-02-09 19:04:07 +00:00
Sebastian Dröge 4d9b6c5472 fmp4mux: Pass one more buffer in test_buffer_multi_stream_short_gops test
This works around non-determinism in aggregator where depending on
timing it can happen that it consumes all buffers from both pads or
waits for another buffer on one pad while the other one already has one.

The effect in this test was that it sometimes timed out. By providing
one more buffer it is guaranteed now that at this point the muxer is
beyond the end of the first fragment.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1081>
2023-02-09 20:36:44 +02:00
Sebastian Dröge 5965ff4364 fmp4mux: Accept more data on already filled streams if the remaining streams need more data for finishing a GOP
In other words, continue queueing buffers in sync from all streams until
all of them are ready for draining instead of stopping to queue buffers
on every stream that is already filled individually.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/310

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1081>
2023-02-09 20:36:42 +02:00
Jan Alexander Steffens (heftig) f55c32ed37 livesync: Document State's fields
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1083>
2023-02-09 13:07:33 +01:00
Jan Alexander Steffens (heftig) 953773a314 livesync: Improve formatting
Move some code around to make it a bit more readable. No change in
behavior.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1083>
2023-02-09 13:07:33 +01:00
Jan Alexander Steffens (heftig) c1bfeb4c23 livesync: Fix log message capitalization
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1083>
2023-02-09 13:07:33 +01:00
Jan Alexander Steffens (heftig) 0af7151ae9 livesync: Extract LiveSync::flow_error
And add details so it behaves more like the `GST_ELEMENT_FLOW_ERROR`
macro.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1083>
2023-02-09 13:07:32 +01:00
Jan Alexander Steffens (heftig) f03ee95bf0 livesync: Extract audio_info_from_caps
And adjust it slightly so it never panics.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1083>
2023-02-09 13:07:32 +01:00
Jan Alexander Steffens (heftig) c971c4d1d5 livesync: Move single segment prop
Keep it with the settings, not after the stats.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1083>
2023-02-09 13:07:32 +01:00
Jan Alexander Steffens (heftig) 165b5f8c50 livesync: Fix queueing
The logic of the element requires the next buffer to be available
immediately after we are done pushing the previous, otherwise we insert
a repeat.

Making the src loop handle events and queries broke this, as upstream is
almost guaranteed not to deliver a buffer in time if we allow non-buffer
items to block upstream's push.

To fix this, replace our single-item `Option` with a `VecDeque` that we
allow to hold an unlimited number of events or queries, but only one
buffer at a time.

In addition, the code was confused about the current caps and segment.

This wasn't an issue before making the src loop handle events and
queries, as only the sinkpad cared about the current segment, using it
to buffers received, and only the srcpad cared about the current caps,
sending it just before sending the next received buffer.

Now the sinkpad cares about caps (through `update_fallback_duration`)
and the srcpad cares about the segment (when not in single-segment
mode).

Fix this by
  - making `in_caps` always hold the current caps of the sinkpad,
  - adding `pending_caps`, which is used by the srcpad to store
    caps to be sent with the next received buffer,
  - adding `in_segment`, holding the current segment of the sinkpad,
  - adding `pending_segment`, which is used by the srcpad to store
    the segment to be sent with the next received buffer,
  - adding `out_segment`, holding the current segment of the srcpad.

Maybe a fix for
https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/298.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1082>
2023-02-09 12:44:47 +01:00
Simon Himmelbauer 3c31c98d95 spotifyaudiosrc: Support configurable bitrate
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1073>
2023-02-09 00:02:30 +02:00
rajneeshksoni 0f383a6545 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/1070>
2023-02-08 14:04:46 +00:00