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Author SHA1 Message Date
Sebastian Dröge
24003a79f6 mpegtslivesrc: Make sure to use the object as context for all debug logs 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1767>
 2024-09-09 13:29:14 +00:00
Sebastian Dröge
c32cb20906 mpegtslivesrc: Check if old compared to new PCR clock estimation is too far off 
It the difference between the two estimations is more than 1s then
consider this a discontinuity too.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1767>
 2024-09-09 13:29:14 +00:00
Sebastian Dröge
c5b1ebc7d8 mpegtslivesrc: Fix order of parameters passed to add_observation() 
The first one should be the internal time, i.e. the monotonic clock time
in our case, and the second one the external time.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1767>
 2024-09-09 13:29:14 +00:00
Sebastian Dröge
44f64fb3f6 mpegtslivesrc: Scale monotonic time on PCR disconts to allow for continuous clock times 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1767>
 2024-09-09 13:29:14 +00:00
Sebastian Dröge
453b3014e6 mpegtslivesrc: Set DISCONT flag on buffers at PCR discontinuities 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1767>
 2024-09-09 13:29:14 +00:00
Sebastian Dröge
8fd2b7daa4 uriplaylistbin: Properly check for stream-collection/streams-selected events 
There is only a new stream-collection event if the media has actually
changed, and that's also the only time in these tests when a
streams-selected event is sent as the default stream selection is always
used and never changed.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/597

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1769>
 2024-09-09 14:37:48 +03:00
Sebastian Dröge
2e0f52de2c uriplaylistbin: Track actual caller in test assertion functions 
Makes it easier to figure out from where it is actually called when the
assertion is failing.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1769>
 2024-09-09 14:35:40 +03:00
Sebastian Dröge
7a959b9e9d uriplaylistbin: Properly clean up the pipeline on panics/assertions too 
Reduces log noise.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1769>
 2024-09-09 14:35:10 +03:00
Sebastian Dröge
a709eb96d9 Fix new Rust 1.81 clippy warnings 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1768>
 2024-09-05 22:01:01 +03:00
Sebastian Dröge
295b9f01c2 ndisrc: Use correct receive time to re-initialize time tracking on disconts 
The base receive time should not be the monotonic system clock time, but
the monotonic system clock time adjusted by the current clock calibration.
For the first time this is equivalent as the clock calibration is the default,
but for further discontinuities it is not and would cause a
discontinuity in the clock times at this point.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1766>
 2024-09-05 10:18:48 +00:00
Mathieu Duponchelle
bfc32cc692 net/aws: fix spurious dispatch failures 
Since https://github.com/awslabs/aws-sdk-rust/discussions/956, the AWS
SDK errors out HTTP streams that do not transfer data for more than 5
seconds.

This probably should be an opt-in bhevior as it clearly not generically
useful, but as it is we need to opt out.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1760>
 2024-09-05 07:43:23 +00:00
Mathieu Duponchelle
65508cfe75 net/aws: don't discard errors from transcribe loop 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1760>
 2024-09-05 07:43:23 +00:00
Arun Raghavan
e72db57179 webrtc: Fix whipclientsink name in README 
The element name was changed, but the documentation wasn't updated to
match.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1764>
 2024-09-03 16:44:19 -04:00
Sebastian Dröge
871756bb70 ndisrc: Reset timestamp tracking if remote time goes backwards 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1727>
 2024-08-29 20:53:13 +03:00
Sebastian Dröge
ee4416ee5f ndisrc: Add a clocked timestamp mode that provides a clock that follows the remote timecodes 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1727>
 2024-08-29 20:53:13 +03:00
Sebastian Dröge
ab3db748be ndisrc: Get rid of unnecessary AtomicRefCell dependency 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1727>
 2024-08-29 16:32:51 +00:00
Sebastian Dröge
0c4ec370cf ndisrc: Remove slope workaround in timestamping code 
This was needed for an old version of the NDI HX Camera iOS application
and is fixed since quite a while. Let's get rid of unnecessarily
complicated code.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1727>
 2024-08-29 16:32:51 +00:00
Sebastian Dröge
57821cade4 ndisrc: Only calculate timecode/timestamp mappings if necessary 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1727>
 2024-08-29 16:32:51 +00:00
Sebastian Dröge
04da3b2047 ndisrc: receiver: Improve debug message when receiving frames 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1727>
 2024-08-29 16:32:51 +00:00
Sebastian Dröge
84fef267b5 ndisrc: receiver: Remove some code duplication 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1727>
 2024-08-29 16:32:51 +00:00
Sebastian Dröge
f2658eb773 ndisrc: Move from start/stop to change_state for slight code simplification 
All state change related code is in a single place now.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1727>
 2024-08-29 16:32:51 +00:00
Seungha Yang
0135c4251c transcriberbin: Fix passthrough state change 
Sync state of child bin appropriately when passthrough is disabled

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1758>
 2024-08-28 05:51:26 +00:00
Sebastian Dröge
c57b6c9c0a Update CHANGELOG.md for 0.13.1 2024-08-27 22:00:48 +03:00
Sebastian Dröge
3c38ed7ff0 deny: Update 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1757>
 2024-08-27 21:10:48 +03:00
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