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Author SHA1 Message Date
Sanchayan Maity
a3e30b499f aws: Introduce a property to use path-style addressing 
AWS SDK switched to virtual addressing as default instead of path
style earlier. While MinIO supports virtual host style requests,
path style requests are the default.

Introduce a property to allow the use of path style addressing if
required.

For more information, see
https://github.com/minio/minio/blob/master/docs/config/README.md#domain
https://docs.aws.amazon.com/AmazonS3/latest/userguide/VirtualHosting.html

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1527>
 2024-04-10 00:23:22 +00:00
François Laignel
2ad452ee89 webrtcsink: don't panic with bitrate handling unsupported encoders 
When an encoder was not supported by the `VideoEncoder` `bitrate` accessors, an
`unimplemented` panic would occur which would poison `state` & `settings`
`Mutex`s resulting in other threads panicking, notably entering `end_session()`,
which lead to many failures in `BinImplExt::parent_remove_element()` until a
segmentation fault ended the process. This was observed using `vaapivp9enc`.

This commit logs a warning if an encoder isn't supported by the `bitrate`
accessors and silently by-passes `bitrate`-related operations when unsupported.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1534>
 2024-04-09 15:48:59 +00:00
Simonas Kazlauskas
5d939498f1 mp4/fmp4: support flac inside the iso (f)mp4 container 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1401>
 2024-04-09 14:37:05 +03:00
Taruntej Kanakamalla
f4b086738b webrtcsrc: change the producer-id type for request-encoded-filter 
With https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1477
the producer id used while emitting the request-encoded-filter
can be a None if the msid of the webrtcbin's pad is None.
This might not affect the signal handler written in C but
can panic in an existing Rust application with signal
handler which can only handle valid String type as its param
for the producer id.

So change the param type to Option<String> in the signal builder
for request-encoded-fiter signal

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1528>
 2024-04-09 06:01:15 +00:00
Tim-Philipp Müller
6b30266145 ci: tag linter and sanity check jobs as a "placeholder" jobs 
They hardly use any resources and almost finish immediately.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1533>
 2024-04-08 23:42:23 +00:00
Tim-Philipp Müller
c8180e714e ci: make sure version Cargo.toml matches version in meson.build 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1532>
 2024-04-08 14:46:00 +01:00
Sebastian Dröge
0b356ee203 deny: Update 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1530>
 2024-04-06 11:12:16 +03:00
Sebastian Dröge
c2ebb3083a Update Cargo.lock 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1530>
 2024-04-06 11:12:16 +03:00
Sebastian Dröge
921938fd20 fmp4mux: Require gstreamer-pbutils 1.20 for the examples 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1530>
 2024-04-06 11:10:58 +03:00
Sebastian Dröge
fab246f82e webrtchttp: Update to reqwest 0.12 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1530>
 2024-04-06 11:07:16 +03:00
Sebastian Dröge
7757e06e36 onvifmetadataparse: Reset state in PAUSED->READY after pad deactivation 
Otherwise the clock id will simply be overridden instead of unscheduling
it, and if the streaming thread of the source pad currently waits on it
then it will wait potentially for a very long time and deactivating the
pad would wait for that to happen.

Also unschedule the clock id on `Drop` of the state to be one the safe
side and not simply forget about it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1526>
 2024-04-05 15:19:37 +00:00
Taruntej Kanakamalla
70adedb142 net/webrtc: fix inconsistencies in documentation of object names 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1529>
 2024-04-05 14:10:35 +00:00
Matthew Waters
7f6929b98d closedcaption: remove libcaption code entirely 
It is now unused.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1517>
 2024-04-05 19:29:24 +11:00
Matthew Waters
2575013faa cea608tott: use our own CEA-608 frame handling instead of libcaption 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1517>
 2024-04-05 19:29:24 +11:00
Matthew Waters
d8fe1c64f1 cea608overlay: use or own CEA-608 caption frame handling instead of libcaption 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1517>
 2024-04-05 19:29:24 +11:00
Matthew Waters
fea85ff9c8 closedcaption: use cea608-types for parsing 608 captions instead of libcaption 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1517>
 2024-04-05 19:29:24 +11:00
François Laignel
cc43935036 webrtc: add precise synchronization example 
This example demonstrates a sender / receiver setup which ensures precise
synchronization of multiple streams in a single session.

[RFC 6051]-style rapid synchronization of RTP streams is available as an option.
See the [Instantaneous RTP synchronization...] blog post for details about this
mode and an example based on RTSP instead of WebRTC.

[RFC 6051]: https://datatracker.ietf.org/doc/html/rfc6051
[Instantaneous RTP synchronization...]: https://coaxion.net/blog/2022/05/instantaneous-rtp-synchronization-retrieval-of-absolute-sender-clock-times-with-gstreamer/

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1463>
 2024-04-03 19:10:40 +02:00
Guillaume Desmottes
b5cbc47cf7 web: webrtcsink: improve panic message on unexpected caps during discovery 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1524>
 2024-04-02 14:25:58 +02:00
Guillaume Desmottes
35b84d219f webrtc: webrtcsink: set perfect-timestamp=true on audio encoders 
Chrome audio decoder doesn't cope well with not perfect ts, generating
noises in the audio.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1524>
 2024-04-02 14:25:51 +02:00
Sebastian Dröge
0aabbb3186 fmp4: Update to dash-mpd 0.16 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1523>
 2024-03-31 09:36:53 +03:00
Sebastian Dröge
4dd6b102c4 Update Cargo.lock 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1523>
 2024-03-31 09:35:46 +03:00
Sebastian Dröge
0dd03da91f ci: Ignore env_logger for cargo-outdated 
It requires Rust >= 1.71.
 2024-03-29 11:03:04 +02:00
Matthew Waters
e1cd52178e transcriberbin: also support 608 inside 708 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1406>
 2024-03-28 13:46:28 +11:00
Matthew Waters
55b4de779c tttocea708: add support for writing 608 compatibility bytes 
608 compatibility bytes are generated using the same functionality as
tttocea608.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1406>
 2024-03-28 13:46:28 +11:00
Matthew Waters
9db4290d2d tttocea608: move functionality to a separate object 
Will be used by tttocea708 later.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1406>
 2024-03-28 13:46:28 +11:00
Matthew Waters
df30d2fbd3 transcriberbin: add support for generating cea708 captions 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1406>
 2024-03-28 13:46:28 +11:00
Matthew Waters
b0cf7e5c81 cea708mux: add element muxing multiple 708 caption services together 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1406>
 2024-03-28 13:46:28 +11:00
Matthew Waters
756abbf807 tttocea708: add element converting from text to cea708 captions 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1406>
 2024-03-28 13:46:28 +11:00
Martin Nordholts
5d7e068a8b rtpgccbwe: Add increasing_duration and counter to existing gst::log!() 
Add `self.increasing_duration` and `self.increasing_counter`
to logs to provide more details of why `overuse_filter()`
determines overuse of network.

To get access to the latest values of those fields we need
to move down the log call. But that is fine, since no other
logged data is modified between the old and new location of
`gst::log!()`.

We do not bother logging `self.last_overuse_estimate` since
that is simply the previously logged value of `estimate`. We
must put the log call before we write the latest value to it
though, in case we want to log it in the future.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1522>
 2024-03-27 15:08:23 +00:00
François Laignel
a870d60621 aws: improve error message logs 
The `Display` and `Debug` trait for the AWS error messages are not very useful.

- `Display` only prints the high level error, e.g.: "service error".
- `Debug` prints all the fields in the error stack, resulting in hard to read
  messages with redudant or unnecessary information. E.g.:

> ServiceError(ServiceError { source: BadRequestException(BadRequestException {
> message: Some("1 validation error detected: Value 'test' at 'languageCode'
> failed to satisfy constraint: Member must satisfy enum value set: [ar-AE,
> zh-HK, en-US, ar-SA, zh-CN, fi-FI, pl-PL, no-NO, nl-NL, pt-PT, es-ES, th-TH,
> de-DE, it-IT, fr-FR, ko-KR, hi-IN, en-AU, pt-BR, sv-SE, ja-JP, ca-ES, es-US,
> fr-CA, en-GB]"), meta: ErrorMetadata { code: Some("BadRequestException"),
> message: Some("1 validation error detected: Value 'test' at 'languageCode'
> failed to satisfy constraint: Member must satisfy enum value set: [ar-AE,
> zh-HK, en-US, ar-SA, zh-CN, fi-FI, pl-PL, no-NO, nl-NL, pt-PT, es-ES, th-TH,
> de-DE, it-IT, fr-FR, ko-KR, hi-IN, en-AU, pt-BR, sv-SE, ja-JP, ca-ES, es-US,
> fr-CA, en-GB]"), extras: Some({"aws_request_id": "1b8bbafd-5b71-4ba5-8676-28432381e6a9"}) } }),
> raw: Response { status: StatusCode(400), headers: Headers { headers:
> {"x-amzn-requestid": HeaderValue { _private: H0("1b8bbafd-5b71-4ba5-8676-28432381e6a9") },
> "x-amzn-errortype": HeaderValue { _private:
> H0("BadRequestException:http://internal.amazon.com/coral/com.amazonaws.transcribe.streaming/") },
> "date": HeaderValue { _private: H0("Tue, 26 Mar 2024 17:41:31 GMT") },
> "content-type": HeaderValue { _private: H0("application/x-amz-json-1.1") },
> "content-length": HeaderValue { _private: H0("315") }} }, body: SdkBody {
> inner: Once(Some(b"{\"Message\":\"1 validation error detected: Value 'test'
> at 'languageCode' failed to satisfy constraint: Member must satisfy enum value
> set: [ar-AE, zh-HK, en-US, ar-SA, zh-CN, fi-FI, pl-PL, no-NO, nl-NL, pt-PT,
> es-ES, th-TH, de-DE, it-IT, fr-FR, ko-KR, hi-IN, en-AU, pt-BR, sv-SE, ja-JP,
> ca-ES, es-US, fr-CA, en-GB]\"}")), retryable: true }, extensions: Extensions {
> extensions_02x: Extensions, extensions_1x: Extensions } } })

This commit adopts the most informative and concise solution I could come up
with to log AWS errors. With the above error case, this results in:

> service error: Error { code: "BadRequestException", message: "1 validation
> error detected: Value 'test' at 'languageCode' failed to satisfy constraint:
> Member must satisfy enum value set: [ar-AE, zh-HK, en-US, ar-SA, zh-CN, fi-FI,
> pl-PL, no-NO, nl-NL, pt-PT, es-ES, th-TH, de-DE, it-IT, fr-FR, ko-KR, hi-IN,
> en-AU, pt-BR, sv-SE, ja-JP, ca-ES, es-US, fr-CA, en-GB]",
> aws_request_id: "a40a32a8-7b0b-4228-a348-f8502087a9f0" }

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1521>
 2024-03-26 20:05:32 +01:00
François Laignel
9f27bde36a aws: use fixed BehaviorVersion 
Quoting [`BehaviorVersion` documentation]:

> Over time, new best-practice behaviors are introduced. However, these
> behaviors might not be backwards compatible. For example, a change which
> introduces new default timeouts or a new retry-mode for all operations might
> be the ideal behavior but could break existing applications.

This commit uses `BehaviorVersion::v2023_11_09()`, which is the latest
major version at the moment. When a new major version is released, the method
will be deprecated, which will warn us of the new version and let us decide
when to upgrade, after any changes if required. This is safer that using
`latest()` which would silently use a different major version, possibly
breaking existing code.

[`BehaviorVersion` documentation]: https://docs.rs/aws-config/1.1.8/aws_config/struct.BehaviorVersion.html

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1520>
 2024-03-26 17:44:16 +01:00
Matthew Waters
e868f81189 gopbuffer: implement element buffering of an entire GOP 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1349>
 2024-03-26 15:29:48 +11:00
Sebastian Dröge
bac2e02160 deny: Add overrides for duplicates hyper / reqwest dependencies 2024-03-24 11:30:30 +02:00
Nirbheek Chauhan
ae7c68dbf8 ci: Add a job to trigger a cerbero build, similar to the monorepo 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1513>
 2024-03-23 23:02:27 +00:00
Sebastian Dröge
0b11209674 Update Cargo.lock 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1510>
 2024-03-23 14:33:07 +02:00
Sebastian Dröge
f97150aa58 reqwest: Update to reqwest 0.12 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1510>
 2024-03-23 14:30:31 +02:00
Philippe Normand
7e1ab086de dav1d: Require dav1d-rs 0.10 
This version depends on libdav1d >= 1.3.0. Older versions are no longer
supported, due to an ABI/API break introduced in 1.3.0.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1512>
 2024-03-21 17:33:32 +00:00
Philippe Normand
be12c0a5f7 Fix clippy warnings after upgrade to Rust 1.77 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1512>
 2024-03-21 17:33:32 +00:00
Sebastian Dröge
317f46ad97 Update CHANGELOG.md for 0.12.3 2024-03-21 18:55:29 +02:00
François Laignel
c5e7e76e4d webrtcsrc: add do-retransmission property 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1509>
 2024-03-21 07:25:30 +00:00
Sebastian Dröge
6556d31ab8 livesync: Ignore another racy test 
Same problem as https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/328
 2024-03-21 09:27:09 +02:00
François Laignel
5476e3d759 webrtcsink: prevent video-info error log for audio streams 
The following error is logged when `webrtcsink` is feeded with an audio stream:

> ERROR video-info video-info.c:540:gst_video_info_from_caps:
>       wrong name 'audio/x-raw', expected video/ or image/

This commit bypasses `VideoInfo::from_caps` for audio streams.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1511>
 2024-03-20 19:57:45 +01:00
François Laignel
cc7b7d508d rtp: gccbwe: don't break downstream assumptions pushing buffer lists 
Some elements in the RTP stack assume all buffers in a `gst::BufferList`
correspond to the same timestamp. See in [`rtpsession`] for instance.
This also had the effect that `rtpsession` did not create correct RTCP as it
only saw some of the SSRCs in the stream.

`rtpgccbwe` formed a packet group by gathering buffers in a `gst::BufferList`,
regardless of whether they corresponded to the same timestamp, which broke
synchronization under certain circonstances.

This commit makes `rtpgccbwe` push the buffers as they were received: one by one.

[`rtpsession`]: bc858976db/subprojects/gst-plugins-good/gst/rtpmanager/gstrtpsession.c (L2462)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1502>
 2024-03-20 18:19:14 +00:00
Sebastian Dröge
2b9272c7eb fmp4mux: Move away from deprecated chrono function 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1503>
 2024-03-20 15:37:18 +02:00
Sebastian Dröge
cca3ebf520 rtp: Switch from chrono to time 
Which allows to simplify quite a bit of code and avoids us having to
handle some API deprecations.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1503>
 2024-03-20 15:05:39 +02:00
Sebastian Dröge
428f670753 version-helper: Use non-deprecated type alias from toml_edit 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1503>
 2024-03-19 18:16:42 +02:00
Sebastian Dröge
fadb7d0a26 deny: Add override for heck 0.4 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1503>
 2024-03-19 17:52:32 +02:00
Sebastian Dröge
2a88e29454 originalbufferstore: Update for VideoMetaTransform -> VideoMetaTransformScale rename 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1503>
 2024-03-19 17:51:41 +02:00
Sebastian Dröge
bfff0f7d87 Update Cargo.lock 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1503>
 2024-03-19 17:50:32 +02:00
Guillaume Desmottes
96337d5234 webrtc: allow resolution and framerate input changes 
Some changes do not require a WebRTC renegotiation so we can allow
those.

Fix #515

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1498>
 2024-03-18 14:52:01 +01:00