Commit graph

372 commits

Author SHA1 Message Date
Sebastian Dröge
16b917abb1 Update for gst::Rank API changes 2023-11-02 14:10:59 +02:00
Sebastian Dröge
855b03a9ea Use let-else instead of match for weak reference upgrades
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1375>
2023-10-30 11:34:35 +02:00
Jan Alexander Steffens (heftig)
e3e58ac0be livesync: Remove the stop from outgoing segments
Our buffer duplication can extend a segment indefinitely.

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/452
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1372>
2023-10-25 19:34:47 +02:00
Jan Alexander Steffens (heftig)
f1ba498b52 livesync: Keep existing buffer duration in some cases
Resize a repeat buffer only if caps gave us a duration to use, or we
consider its current duration unreasonable.

In particular, for audio streams we should prefer reusing the buffer
size upstream gave us, as we did before 6633cc4046.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1372>
2023-10-25 19:34:47 +02:00
Jan Alexander Steffens (heftig)
59beade079 livesync: Split fallback_duration into in_ and out_duration
Make it independent of the `latency`; this was inconsistent anyway,
where the default latency of zero got you a fallback duration of 100 ms
and something else got you half the latency.

Maintain a separate duration for the `in` and the `out` side so we
change the duration of repeat buffers after a caps change, not just
before.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1372>
2023-10-25 19:08:16 +02:00
Guillaume Desmottes
f94ecfc7a6 livesync: display jitter when waiting on clock
We already log the result of the clock wait call so may as well log the
returned jitter.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1370>
2023-10-25 14:26:19 +02:00
Guillaume Desmottes
13dae0f0d0 livesync: log new pending segments
The debug print of the event does not display details about the segment:
  Unqueueing Some(Event(Event { ptr: 0x7fa3e0002580, type: "segment", seqnum: Seqnum(479), structure: Some(GstEventSegment { segment: (GstSegment) ((GstSegment*) 0x7fa3e8001d00) }) }))

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1370>
2023-10-25 14:24:35 +02:00
Jan Alexander Steffens (heftig)
ee93448de7 livesync: example: Add identities single-segment=1
These let us change the runtime offset of the test buffers via pad
offsets without pushing new segments into livesync, which is necessary
to demo the late-threshold behavior.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1369>
2023-10-25 12:55:06 +02:00
Jan Alexander Steffens (heftig)
6633cc4046 livesync: Use fallback_duration for audio repeat buffers as well
Don't depend on upstream giving us sanely-sized buffers if we want to
repeat.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1369>
2023-10-25 12:55:06 +02:00
Jan Alexander Steffens (heftig)
4ac7d0415b livesync: Separate out_buffer duplicate status from GAP flag
Otherwise we might get confused by upstream GAP buffers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1369>
2023-10-25 12:10:40 +02:00
Jan Alexander Steffens (heftig)
2f36bd5d77 livesync: Handle flags and late buffer patching after queueing
This makes the chain function almost independent of the output state. We
still do the early discard check with `buffer_is_backwards` so we don't
try to queue buffers we can't use, allowing us to fast-forward upstream
without blocking on the src task.

Don't accept `LateOverThreshold` buffers when we have `pending_caps` or
a `pending_segment`. We need to apply these first before we can sensibly
patch buffers from the new stream.

Deduplicate most of the output buffer patching code into a new
`patch_output_buffer` method.

For: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/450
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1369>
2023-10-25 11:52:41 +02:00
Jan Alexander Steffens (heftig)
7c48a299c3 livesync: Simplify num_duplicate counting
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1369>
2023-10-25 11:52:40 +02:00
Jan Alexander Steffens (heftig)
17a2448237 livesync: Move num_in counting to the src task
This is in preparation for moving more accept/discard logic to the src
task, so we can only count `num_in` here.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1369>
2023-10-25 11:52:40 +02:00
Jan Alexander Steffens (heftig)
1740a8e363 livesync: Move a notify closer to the interesting state change
Move the `notify_all` to where we pop the buffer. We're moving within a
single state lock so no change in behavior.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1369>
2023-10-25 11:52:40 +02:00
Jan Alexander Steffens (heftig)
44f2195674 livesync: Replace an if-let with match
No change in behavior, yet. Separate commit to ease reviewing.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1369>
2023-10-25 11:52:40 +02:00
Jan Alexander Steffens (heftig)
62791bfb47 livesync: Clean up state handling
- Separate resetting state more cleanly, introducing `set_flushing`,
  `sink_reset` and `src_reset`.
- Clear the queue early when we flush, in order to unblock waits on
  query responses.
- Return an error when we fail to start, pause or stop the task.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1369>
2023-10-25 11:52:40 +02:00
Jan Alexander Steffens (heftig)
d663f708ef livesync: Log a category error when we are missing the segment
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1369>
2023-10-25 11:52:40 +02:00
Jan Alexander Steffens (heftig)
6567041a3d livesync: Improve audio duration fixups
- An entirely missing duration is now only logged at debug level instead
  of pretending the duration was zero and warning about it.
- Silently fix up a duration difference up to one sample.
- Error when we fail to calculate the duration; don't try to apply the
  `fallback_duration` to a non-video stream.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1369>
2023-10-25 11:52:40 +02:00
Jan Alexander Steffens (heftig)
0a45f776e0 livesync: Simplify start_src_task and src_loop
This should effect no change in behavior.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1369>
2023-10-25 11:03:15 +02:00
Jan Alexander Steffens (heftig)
01386b8451 livesync: Rename activatemode methods to *_activatemode
This matches the other plugins.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1369>
2023-10-25 11:03:14 +02:00
Sebastian Dröge
d468e1e4a6 Clean up usage of pad probes
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1361>
2023-10-17 08:44:06 +03:00
Guillaume Desmottes
f92dc28696 fallbackswitch: protect src pad stream lock using Cond
Should prevent stream and State deadlocks, see https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/202

Fix #202
Hopefully fix #192 as well.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1300>
2023-08-14 14:18:41 +02:00
Guillaume Desmottes
d3da30be6d fallbackswitch: prevent deadlocks in chain function
Calling schedule_timeout() may result in handle_timeout() being called right away,
which will need pad state locks which was still hold in chain().

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1300>
2023-08-14 13:16:37 +02:00
Guillaume Desmottes
100333c021 fallbackswitch: ensure strict ordering when taking mutexes
Should prevent deadlocks.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1300>
2023-08-14 13:16:37 +02:00
Sebastian Dröge
d688aeb184 Update versions to 0.12.0-alpha.1 2023-08-10 17:21:11 +03:00
Sebastian Dröge
3b41f206bc Don't generate .def files for plugins
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/389

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1299>
2023-08-09 13:54:34 +03:00
François Laignel
10902c0485 utils: fix further to glib change ControlFlow -> Propagation
See: https://github.com/gtk-rs/gtk-rs-core/pull/1144
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1292>
2023-08-02 13:19:22 +02:00
Sebastian Dröge
31b1cb8ca6 Update minimum supported Rust version to 1.70
gtk-rs will update soonish too.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1280>
2023-07-19 09:19:34 +03:00
Sebastian Dröge
bbd3d9ffe0 Remove unnecessary mut everywhere
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1273>
2023-07-11 10:09:35 +03:00
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
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
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
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
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
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
Sebastian Dröge
c65b3429ad Use MPL as license specifier for plugins only requiring GStreamer < 1.20
And use MPL-2.0 for all that require GStreamer 1.20 or newer. The new
string is only allowed in 1.20 or newer and using it in older versions
causes failure to load the plugin.

All affected plugins are of course still MPL-2.0 licensed.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/374

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1235>
2023-06-07 19:13:55 +03:00
Guillaume Desmottes
f4604e1c58 uriplaylistbin: use thiserror
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1232>
2023-06-06 12:46:17 +02:00
Guillaume Desmottes
432de060ea uriplaylistbin: example: display iterations
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1232>
2023-06-05 14:09:41 +02:00
Guillaume Desmottes
97fa20237f uriplaylistbin: prevent deadlock when notifying property changes
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1232>
2023-06-05 14:09:41 +02:00
Guillaume Desmottes
780d9d5b78 uriplaylistbin: example display when leaving because of eos
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1232>
2023-06-05 14:09:41 +02:00
Seungha Yang
3406e604cd fallbacksrc: Don't apply fallback-audio-caps to the main audio stream
Intended behavior is configuring audio convert/resample elements
only for the fallback stream and also fallback-audio-caps is set.
Video and image stream are doing it as intended already.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1213>
2023-05-17 23:49:09 +09:00