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Author SHA1 Message Date
Sebastian Dröge
ee4aca3010 webrtcsink: Set config-interval=-1 and aggregate-mode=zero-latency on rtph26[45]pay 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1272>
 2023-07-10 19:48:37 +03:00
Sebastian Dröge
957a28f239 webrtcsink: Set VP8/VP9 payloader based on payloader element factory name 
Instead of checking the encoder's name. There are more VP8/VP9 encoders
than the ones from the vpx plugin.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1272>
 2023-07-10 19:45:17 +03:00
Sebastian Dröge
1bb06d775c fmp4: Update to dash-mpd 0.11 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1271>
 2023-07-10 08:29:29 +03:00
Sebastian Dröge
b6bc7fb6d5 Revert "cargo-deny: Remove bitflags from the list" 
This reverts commit 64151790d0.
Some dependencies still use bitflags 1
 2023-07-07 09:08:50 +03:00
Mathieu Duponchelle
1dd13c4812 webrtcsink: fix session_id / peer_id confusion 
In a few places, for instance parameter names, peer_id was still used
when session_id was actually getting passed.

Go through all instances of peer_id in webrtcsink/imp.rs and address
those mix-ups.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1269>
 2023-07-07 05:33:30 +00:00
Bilal Elmoussaoui
0fa2c861d6 Adapt to removal of glib::Inhibit 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1270>
 2023-07-06 22:02:31 +02:00
Jan Schmidt
2abc72b606 fallbackswitch: Change the threshold for trailing buffers 
Only discard buffers on inactive pads if they are later
than the current output running time, rather than the
later timeout running time. That can mean switching
to a higher priority pad can happen quicker.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1198>
 2023-07-07 01:45:52 +10:00
Jan Schmidt
61e282af1a fallbackswitch: Fix pad health calculation and notifies 
Change the pad health calculation to consider a pad 'healthy'
if it has received data within the last 'timeout' window. Previously,
inactive pads were constantly flip-flopping between healthy and not
healthy depending on whether they were slightly ahead of or behind
the active pad running_time.

When the health status of a pad changes, make sure to always notify
the property, so that applications that are manually controlling
the active pad can make their switching decisions.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1198>
 2023-07-07 01:45:52 +10:00
Bilal Elmoussaoui
dd2d7d9215 Use re-exported once_cell 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1268>
 2023-07-06 17:50:49 +03:00
Bilal Elmoussaoui
2cc98bf410 Adapt to glib::Continue rename 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1268>
 2023-07-06 17:50:49 +03:00
Bilal Elmoussaoui
64151790d0 cargo-deny: Remove bitflags from the list 
We now use bitflags 2.x

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1268>
 2023-07-06 15:26:13 +02:00
Sebastian Dröge
58adebb325 Fix a couple of typos 2023-07-06 13:50:17 +03:00
Sebastian Dröge
983f990fe9 Update docs after GStreamer update on the CI 2023-07-06 13:48:59 +03:00
Sebastian Dröge
4918bf0ab2 deny: Update 2023-07-06 08:55:14 +03:00
Olivier Crête
08b6251a7a webrtc-utils: Ensure there is only one cancellable call at a time 
Since we only have one canceller at a time, panic if one try to
use it twice in parallel.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1262>
 2023-07-05 21:43:17 +00:00
Olivier Crête
817b60a758 webrtc: Value.get() is already type checks in the property calls 
GObject will have ensured we get a GValue of the right type.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1262>
 2023-07-05 21:43:17 +00:00
Olivier Crête
793ee66afa webrtcsink: Add LiveKit WebRTC sink and signaller 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1262>
 2023-07-05 21:43:17 +00:00
Sebastian Dröge
de6abf1439 Update CHANGELOG.md for 0.10.10 2023-07-05 16:03:42 +03:00
Vivia Nikolaidou
9d7af671c5 togglerecord: Clip segment before calculating timestamp/duration 
Clipping happens in buffer time space and data.clip can modify the
buffer timestamp and duration. Move it before we calculate them in order
to make it actually have some effect.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1265>
 2023-07-04 09:07:36 +00:00
yatinmaan
1ed9992775 gtk4: Add python example 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1259>
 2023-07-02 00:31:32 +00:00
Vivia Nikolaidou
8417efc630 togglerecord: Error out if main stream buffer has no valid running time 
We cannot continue with this buffer, because we cannot calculate the
time when the recording stopped or started. We also cannot safely drop
it, because that might break the stream, especially if it's encoded.
Therefore, we return an element error.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1264>
 2023-06-30 09:20:42 +00:00
Sebastian Dröge
cc26f06289 deny: Update 2023-06-30 11:12:52 +03:00
Seungha Yang
1f0ce101eb awstranscriber: Tone down log message 
It's not an ERROR case at all

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1263>
 2023-06-28 23:57:54 +09:00
Sebastian Dröge
4555b98aff deny: Update 2023-06-27 10:58:57 +03:00
Sebastian Dröge
c350f3c2af webrtcink: Use correct property types for nvvideoconvert 
These are enums and not plain integers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1256>
 2023-06-26 14:48:58 +00:00
Vivia Nikolaidou
8366716456 togglerecord: Change test_two_stream_close_open_nonlivein_liveout timeout to 60ms 
20ms was not enough for the CI.

Closes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/379

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1258>
 2023-06-26 12:33:31 +00:00
Sebastian Dröge
effe1bacdf deny: Update 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1260>
 2023-06-26 14:24:08 +03:00
Sebastian Dröge
bbb88da9e0 videofx: Minimize dependencies of the image crate 
Only the basic infrastructure is needed and none of the
decoders/encoders for various image formats.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1260>
 2023-06-26 14:21:44 +03:00
Sebastian Dröge
f481bb74c4 fmp4: Update to dash-mpd 0.10 for the example 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1260>
 2023-06-26 14:19:43 +03:00
Sebastian Dröge
042a297d1a gtk4: Update to windows-sys 0.48 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1260>
 2023-06-26 14:19:22 +03:00
Jayson Reis
e58abf0705 gtk4: Make winegl code compilable 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1255>
 2023-06-23 11:16:40 +03:00
Jayson Reis
d3d78846dc gtk4: Fix code to run on current main branch 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1255>
 2023-06-22 14:45:28 +02:00
Sebastian Dröge
dcb80ac105 gtk4: Add support for GL on Windows 
This implements all the workarounds for Windows-specific complications
that the GTK GStreamer mediafile implementation also does.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1255>
 2023-06-22 07:43:57 +02:00
Vivia Nikolaidou
2be14b95b3 togglerecord: Fix nonlive inputs when element is started not recording 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1252>
 2023-06-21 16:27:00 +03:00
Sebastian Dröge
1119ed6620 livesync: Wait for the end timestamp of the previous buffer before looking at queue 
Previously livesync was waiting for the start timestamp of the current
buffer after looking at the queue and right before pushing it
downstream. This meant that it generally looked too early at the queue
and especially that upstream had to provide the next buffer already at
the start timestamp of the previous one.

Instead, now wait before looking at the queue and wait for the end
timestamp of the previous buffer. Once the previous buffer has expired,
a new buffer really needs to be available or otherwise a filler buffer
has to be pushed downstream.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1250>
 2023-06-20 13:01:39 +00:00
Jan Alexander Steffens (heftig)
52ded6e8cc livesync: Improve EOS handling 
I've looked at the GstQueue code again and tried making livesync behave
better with EOS. This isn't very well tested, though. My goal was to
make this look saner but I think this should be reviewed by someone who
knows the queue code.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1251>
 2023-06-20 13:18:17 +02:00
Sebastian Dröge
99fc5f16d2 Update CHANGELOG.md for 0.10.9 2023-06-19 20:48:22 +03:00
François Laignel
f85106b86a mp4, fmp4: fix byte order for opus extension 
The "Encapsulation of Opus in ISO Base Media File Format" [1] specifications,
§ 4.3.2 Opus Specific Box, indicates that data must be stored as big-endian.

In `write_dops`, `to_le_bytes` variants were used.

Related to [2].

[1] https://opus-codec.org/docs/opus_in_isobmff.html#4.3.2
[2] https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4875

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1247>
 2023-06-19 16:32:07 +02:00
Mathieu Duponchelle
84a33ca7b9 webrtcsink: bring in signalling code from whipsink as a signaller 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1168>
 2023-06-16 00:32:56 +02:00
Mathieu Duponchelle
f00a169081 webrtcsrc: add twcc extension to codec-preferences when present 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1245>
 2023-06-15 20:41:53 +00:00
Seungha Yang
02c77d2e44 mccparse: Map timecode to PTS directly without offset 
Assumes that caption stream's timeline starts from zero,
and maps timecode time_since_daily_jam() to PTS directly without
subtracting the first seen timecode.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1246>
 2023-06-16 01:06:26 +09:00
Stéphane Cerveau
26fd68a37c gitlab: add issue template 
Use the same bug template as in gstreamer repository

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1244>
 2023-06-15 11:01:15 +00:00
Sebastian Dröge
21df8f8c08 deny: Update 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1243>
 2023-06-15 10:18:47 +03:00
Sebastian Dröge
63df653ad2 fmp4mux: Update to quick-xml 0.29 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1243>
 2023-06-15 10:15:52 +03:00
Mathieu Duponchelle
1200ae0ee6 webrtcsink: improve debug 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1239>
 2023-06-14 22:27:15 +02:00
Mathieu Duponchelle
64056c5527 net/webrtc: improve documentation layout 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1239>
 2023-06-14 22:27:15 +02:00
Vivia Nikolaidou
063871a1eb togglerecord: Add support for non-live inputs 
Live input + is-live=false:
    While not recording, drop input
    When recording is started, offset to collapse the gap

Live input + is-live=true:
    While not recording, drop input
    Don't modify the offset

Non-live input + is-live=false:
    While not recording, block input
    Don't modify the offset

Non-live input + is-live=true:
    While not recording, block input
    When recording is started, offset to current running time

Co-authored-by: Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1206>
 2023-06-14 15:58:04 +03:00
Guillaume Desmottes
4683291c1f fallbackswitch: add 'stop-on-eos' property 
Fix the following use case:
- main input of fallbackswitch is finite (a media file)
- fallback input is infinite (videotestsrc)
- main input is done and send eos, which is propagated downstream
- fallbackswitch switches to fallback, sending STREAM_START which reset
  EOS downstream (aggregator does that)
- fallback input keeps pushing buffers forever.

Solve it by adding a 'stop-on-eos' property so fallbackswitch stops
pushing property once the main input is eos.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1242>
 2023-06-13 14:49:06 +02:00
Guillaume Desmottes
6ad0db2cdb fallbackswitch: remove unused SinkState::eos 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1242>
 2023-06-13 12:43:51 +02:00
Guillaume Desmottes
692d1bfb9e fallbackswitch: log when handling events 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1242>
 2023-06-13 12:43:51 +02:00