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gst-plugins-rs/docs/plugins/gst_plugins_cache.json

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Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
{
"aws": {
Update docs
2022-10-23 17:35:14 +00:00
"description": "GStreamer Amazon Web Services plugin",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"elements": {
"awss3hlssink": {
"author": "Daily. Co",
"description": "Streams HLS data to S3",
"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstAwsS3HlsSink",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"GstBin",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"interfaces": [
"GstChildProxy"
],
"klass": "Generic",
"long-name": "S3 HLS Sink",
"pad-templates": {
"audio": {
"caps": "ANY",
"direction": "sink",
"presence": "request"
},
"video": {
"caps": "ANY",
"direction": "sink",
"presence": "request"
}
},
"properties": {
"access-key": {
"blurb": "AWS Access Key",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
},
"acl": {
"blurb": "Canned ACL to use for uploading to S3",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "private",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
"bucket": {
"blurb": "The bucket of the file to write",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
},
"endpoint-uri": {
"blurb": "The S3 endpoint URI to use",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
},
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>
2024-04-05 08:13:10 +00:00
"force-path-style": {
"blurb": "Force client to use path-style addressing for buckets",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "null",
"readable": true,
"type": "gboolean",
"writable": true
},
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"hlssink": {
"blurb": "The underlying HLS sink being used",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"mutable": "null",
"readable": true,
"type": "GstElement",
"writable": false
},
"key-prefix": {
"blurb": "The key prefix for segment and playlist files",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
},
"region": {
"blurb": "The AWS region for the S3 bucket (e.g. eu-west-2).",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "us-west-2",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
},
"request-timeout": {
"blurb": "Timeout for request to S3 service (in ms)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "15000",
"max": "18446744073709551615",
"min": "1",
"mutable": "null",
"readable": true,
"type": "guint64",
"writable": true
},
"retry-attempts": {
"blurb": "Number of times AWS SDK attempts a request before abandoning the request",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "5",
"max": "10",
"min": "1",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
"secret-access-key": {
"blurb": "AWS Secret Access Key",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
},
"session-token": {
"blurb": "AWS temporary session token from STS",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
awss3hlssink: Add stats property. application can monitor the progress of hls segment generation and upload progress. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1022>
2022-12-19 12:46:54 +00:00
},
"stats": {
"blurb": "Various statistics",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "stats, num-uploads-started=(int)0, num-uploads-completed=(int)0, num-bytes-uploaded=(int)0;",
"mutable": "null",
"readable": true,
"type": "GstStructure",
"writable": false
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
}
},
"rank": "none"
},
aws: Add plugin docs Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1337>
2023-12-18 20:38:36 +00:00
"awss3putobjectsink": {
"author": "Arun Raghavan <arun@asymptotic.io>",
"description": "Writes an object to Amazon S3 using PutObject (mostly useful for small files)",
"hierarchy": [
"GstAwsS3PutObjectSink",
"GstBaseSink",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Source/Network",
"pad-templates": {
"sink": {
"caps": "ANY",
"direction": "sink",
"presence": "always"
}
},
"properties": {
"access-key": {
"blurb": "AWS Access Key",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
},
"bucket": {
"blurb": "The bucket of the file to write",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
},
aws: Add system-defined metadata options to both sinks Add to awss3sink and awss3putobjectsink elements the following paramerters which are set on the uploaded S3 objects: * cache-control; * content-encoding; and * content-language Bugfix: Set the content-type and content-disposition values in the S3 putobject call. Previously the params were defined on the element but unused. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1585>
2024-05-24 10:11:19 +00:00
"cache-control": {
"blurb": "Cache-Control header to set for uploaded object",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
aws: Add plugin docs Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1337>
2023-12-18 20:38:36 +00:00
"content-disposition": {
"blurb": "Content-Disposition header to set for uploaded object",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
aws: Add system-defined metadata options to both sinks Add to awss3sink and awss3putobjectsink elements the following paramerters which are set on the uploaded S3 objects: * cache-control; * content-encoding; and * content-language Bugfix: Set the content-type and content-disposition values in the S3 putobject call. Previously the params were defined on the element but unused. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1585>
2024-05-24 10:11:19 +00:00
"content-encoding": {
"blurb": "Content-Encoding header to set for uploaded object",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
"content-language": {
"blurb": "Content-Language header to set for uploaded object",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
aws: Add plugin docs Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1337>
2023-12-18 20:38:36 +00:00
"content-type": {
"blurb": "Content-Type header to set for uploaded object",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
"endpoint-uri": {
"blurb": "The S3 endpoint URI to use",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
"flush-interval-buffers": {
"blurb": "Number of buffers to accumulate before doing a write (0 => disable)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "1",
"max": "18446744073709551615",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint64",
"writable": true
},
"flush-interval-bytes": {
"blurb": "Number of bytes to accumulate before doing a write (0 => disable)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "18446744073709551615",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint64",
"writable": true
},
"flush-interval-time": {
"blurb": "Total duration of buffers to accumulate before doing a write (0 => disable)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "18446744073709551615",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint64",
"writable": true
},
"flush-on-error": {
"blurb": "Whether to write out the data on error (like stopping without an EOS)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "null",
"readable": true,
"type": "gboolean",
"writable": true
},
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>
2024-04-05 08:13:10 +00:00
"force-path-style": {
"blurb": "Force client to use path-style addressing for buckets",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "null",
"readable": true,
"type": "gboolean",
"writable": true
},
aws: Add plugin docs Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1337>
2023-12-18 20:38:36 +00:00
"key": {
"blurb": "The key of the file to write",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
},
"metadata": {
"blurb": "A map of metadata to store with the object in S3; field values need to be convertible to strings.",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"mutable": "ready",
"readable": true,
"type": "GstStructure",
"writable": true
},
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>
2024-04-17 16:11:06 +00:00
"min-keyframe-distance": {
"blurb": "Minimum distance between keyframes to start a new file",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "10000000000",
"max": "18446744073709551615",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint64",
"writable": true
},
"next-file": {
"blurb": "When to start new file",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "next-buffer (0)",
"mutable": "ready",
"readable": true,
"type": "GstS3PutObjectSinkNextFile",
"writable": true
},
aws: Add plugin docs Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1337>
2023-12-18 20:38:36 +00:00
"region": {
"blurb": "An AWS region (e.g. eu-west-2).",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "us-west-2",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
},
"request-timeout": {
"blurb": "Timeout for general S3 requests (in ms, set to -1 for infinity)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "15000",
"max": "9223372036854775807",
"min": "-1",
"mutable": "null",
"readable": true,
"type": "gint64",
"writable": true
},
"retry-attempts": {
"blurb": "Number of times AWS SDK attempts a request before abandoning the request",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "5",
"max": "10",
"min": "1",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
"secret-access-key": {
"blurb": "AWS Secret Access Key",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
},
"session-token": {
"blurb": "AWS temporary Session Token from STS",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
},
"uri": {
"blurb": "The S3 object URI",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
}
},
"rank": "none"
},
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"awss3sink": {
"author": "Marcin Kolny <mkolny@amazon.com>",
"description": "Writes an object to Amazon S3",
"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstAwsS3Sink",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"GstBaseSink",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"interfaces": [
"GstURIHandler"
],
"klass": "Source/Network",
"long-name": "Amazon S3 sink",
"pad-templates": {
"sink": {
"caps": "ANY",
"direction": "sink",
"presence": "always"
}
},
"properties": {
"access-key": {
"blurb": "AWS Access Key",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
},
"bucket": {
"blurb": "The bucket of the file to write",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
},
aws: Add system-defined metadata options to both sinks Add to awss3sink and awss3putobjectsink elements the following paramerters which are set on the uploaded S3 objects: * cache-control; * content-encoding; and * content-language Bugfix: Set the content-type and content-disposition values in the S3 putobject call. Previously the params were defined on the element but unused. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1585>
2024-05-24 10:11:19 +00:00
"cache-control": {
"blurb": "Cache-Control header to set for uploaded object",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"complete-upload-request-timeout": {
"blurb": "Timeout for the complete multipart upload request (in ms, set to -1 for infinity) (Deprecated. Use request-timeout.)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "15000",
"max": "9223372036854775807",
"min": "-1",
"mutable": "null",
"readable": true,
"type": "gint64",
"writable": true
},
"complete-upload-retry-duration": {
"blurb": "How long we should retry complete multipart upload requests before giving up (in ms, set to -1 for infinity) (Deprecated. Use retry-attempts.)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "75000",
"max": "9223372036854775807",
"min": "-1",
"mutable": "null",
"readable": true,
"type": "gint64",
"writable": true
},
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 13:44:16 +00:00
"content-disposition": {
"blurb": "Content-Disposition header to set for uploaded object",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
aws: Add system-defined metadata options to both sinks Add to awss3sink and awss3putobjectsink elements the following paramerters which are set on the uploaded S3 objects: * cache-control; * content-encoding; and * content-language Bugfix: Set the content-type and content-disposition values in the S3 putobject call. Previously the params were defined on the element but unused. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1585>
2024-05-24 10:11:19 +00:00
"content-encoding": {
"blurb": "Content-Encoding header to set for uploaded object",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
"content-language": {
"blurb": "Content-Language header to set for uploaded object",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
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 13:44:16 +00:00
"content-type": {
"blurb": "Content-Type header to set for uploaded object",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"endpoint-uri": {
"blurb": "The S3 endpoint URI to use",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
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>
2024-04-05 08:13:10 +00:00
"force-path-style": {
"blurb": "Force client to use path-style addressing for buckets",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "null",
"readable": true,
"type": "gboolean",
"writable": true
},
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"key": {
"blurb": "The key of the file to write",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
},
"metadata": {
"blurb": "A map of metadata to store with the object in S3; field values need to be convertible to strings.",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"mutable": "ready",
"readable": true,
"type": "GstStructure",
"writable": true
},
"on-error": {
"blurb": "Do nothing, abort or complete a multipart upload request on error",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "nothing (2)",
"mutable": "ready",
"readable": true,
"type": "GstS3SinkOnError",
"writable": true
},
"part-size": {
"blurb": "A size (in bytes) of an individual part used for multipart upload.",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "5242880",
"max": "5368709120",
"min": "5242880",
"mutable": "ready",
"readable": true,
"type": "guint64",
"writable": true
},
"region": {
"blurb": "An AWS region (e.g. eu-west-2).",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "us-west-2",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
},
"request-timeout": {
"blurb": "Timeout for general S3 requests (in ms, set to -1 for infinity)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "15000",
"max": "9223372036854775807",
"min": "-1",
"mutable": "null",
"readable": true,
"type": "gint64",
"writable": true
},
"retry-attempts": {
"blurb": "Number of times AWS SDK attempts a request before abandoning the request",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "5",
"max": "10",
"min": "1",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
"retry-duration": {
"blurb": "How long we should retry general S3 requests before giving up (in ms, set to -1 for infinity) (Deprecated. Use retry-attempts.)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "75000",
"max": "9223372036854775807",
"min": "-1",
"mutable": "null",
"readable": true,
"type": "gint64",
"writable": true
},
"secret-access-key": {
"blurb": "AWS Secret Access Key",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
},
"session-token": {
"blurb": "AWS temporary Session Token from STS",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
},
"upload-part-request-timeout": {
"blurb": "Timeout for a single upload part request (in ms, set to -1 for infinity) (Deprecated. Use request-timeout.)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "15000",
"max": "9223372036854775807",
"min": "-1",
"mutable": "null",
"readable": true,
"type": "gint64",
"writable": true
},
"upload-part-retry-duration": {
"blurb": "How long we should retry upload part requests before giving up (in ms, set to -1 for infinity) (Deprecated. Use retry-attempts.)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "75000",
"max": "9223372036854775807",
"min": "-1",
"mutable": "null",
"readable": true,
"type": "gint64",
"writable": true
},
"uri": {
"blurb": "The S3 object URI",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
}
},
"rank": "primary"
},
"awss3src": {
"author": "Arun Raghavan <arun@arunraghavan.net>",
"description": "Reads an object from Amazon S3",
"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstAwsS3Src",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"GstBaseSrc",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"interfaces": [
"GstURIHandler"
],
"klass": "Source/Network",
"long-name": "Amazon S3 source",
"pad-templates": {
"src": {
"caps": "ANY",
"direction": "src",
"presence": "always"
}
},
"properties": {
"access-key": {
"blurb": "AWS Access Key",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
},
"endpoint-uri": {
"blurb": "The S3 endpoint URI to use",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
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>
2024-04-05 08:13:10 +00:00
"force-path-style": {
"blurb": "Force client to use path-style addressing for buckets",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "null",
"readable": true,
"type": "gboolean",
"writable": true
},
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"request-timeout": {
"blurb": "Timeout for each S3 request (in ms, set to -1 for infinity)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "15000",
"max": "9223372036854775807",
"min": "-1",
"mutable": "null",
"readable": true,
"type": "gint64",
"writable": true
},
"retry-attempts": {
"blurb": "Number of times AWS SDK attempts a request before abandoning the request",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "5",
"max": "10",
"min": "1",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
"retry-duration": {
"blurb": "How long we should retry S3 requests before giving up (in ms, set to -1 for infinity) (Deprecated. Use retry-attempts.)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "75000",
"max": "9223372036854775807",
"min": "-1",
"mutable": "null",
"readable": true,
"type": "gint64",
"writable": true
},
"secret-access-key": {
"blurb": "AWS Secret Access Key",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
},
"session-token": {
"blurb": "AWS temporary Session Token from STS",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
},
"uri": {
"blurb": "The S3 object URI",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
}
},
"rank": "primary"
},
"awstranscribeparse": {
"author": "Mathieu Duponchelle <mathieu@centricular.com>",
"description": "Parses AWS transcripts into timed text buffers",
"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstAwsTranscribeParse",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Text/Subtitle",
"long-name": "AWS transcript parser",
"pad-templates": {
"sink": {
"caps": "application/x-json:\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "text/x-raw:\n format: utf8\n",
"direction": "src",
"presence": "always"
}
},
"rank": "none"
},
"awstranscriber": {
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>
2023-02-24 20:40:54 +00:00
"author": "Jordan Petridis <jordan@centricular.com>, Mathieu Duponchelle <mathieu@centricular.com>, François Laignel <francois@centricular.com>",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"description": "Speech to Text filter, using AWS transcribe",
"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstAwsTranscriber",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
net/aws/transcriber: add translation request src pads This commit adds an optional transcript translation feature implemented as request src Pads. When requesting a src Pad, the user can specify the translation language code using Pad properties 'language-code'. The following properties are defined on the Element: - 'transcribe-latency': formerly 'latency', defines the expected latency for the Transcribe webservice. - 'translate-latency': defines the expected latency for the Translate webservice. - 'transcript-lookahead': maximum transcript duration to send to translation when a transcript is hitting its deadline and no punctuation was found. When the input and output languages are the same, only the 'transcribe-latency' is used for the Pad. Otherwise, the resulting latency is the addition of 'transcribe-latency' and 'translate-latency'. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1109>
2023-03-10 13:47:38 +00:00
"interfaces": [
"GstChildProxy"
],
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"klass": "Audio/Text/Filter",
"long-name": "Transcriber",
"pad-templates": {
"sink": {
plugins, examples, tutorials: Use AudioCapsBuilder and VideoCapsBuilder Simplify caps creation code
2022-10-13 18:02:04 +00:00
"caps": "audio/x-raw:\n rate: [ 8000, 48000 ]\n channels: 1\n layout: { (string)interleaved, (string)non-interleaved }\n format: S16LE\n",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "text/x-raw:\n format: utf8\n",
"direction": "src",
net/aws/transcriber: add translation request src pads This commit adds an optional transcript translation feature implemented as request src Pads. When requesting a src Pad, the user can specify the translation language code using Pad properties 'language-code'. The following properties are defined on the Element: - 'transcribe-latency': formerly 'latency', defines the expected latency for the Transcribe webservice. - 'translate-latency': defines the expected latency for the Translate webservice. - 'transcript-lookahead': maximum transcript duration to send to translation when a transcript is hitting its deadline and no punctuation was found. When the input and output languages are the same, only the 'transcribe-latency' is used for the Pad. Otherwise, the resulting latency is the addition of 'transcribe-latency' and 'translate-latency'. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1109>
2023-03-10 13:47:38 +00:00
"presence": "always",
net/aws/transcriber: rename prop transcript-lookahead & TranslationSrcPad ... as translate-lookahead and TranslateSrcPad. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1137>
2023-03-15 13:59:50 +00:00
"type": "GstTranslateSrcPad"
net/aws/transcriber: add translation request src pads This commit adds an optional transcript translation feature implemented as request src Pads. When requesting a src Pad, the user can specify the translation language code using Pad properties 'language-code'. The following properties are defined on the Element: - 'transcribe-latency': formerly 'latency', defines the expected latency for the Transcribe webservice. - 'translate-latency': defines the expected latency for the Translate webservice. - 'transcript-lookahead': maximum transcript duration to send to translation when a transcript is hitting its deadline and no punctuation was found. When the input and output languages are the same, only the 'transcribe-latency' is used for the Pad. Otherwise, the resulting latency is the addition of 'transcribe-latency' and 'translate-latency'. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1109>
2023-03-10 13:47:38 +00:00
},
"src_%%u": {
"caps": "text/x-raw:\n format: utf8\n",
"direction": "src",
"presence": "request",
net/aws/transcriber: rename prop transcript-lookahead & TranslationSrcPad ... as translate-lookahead and TranslateSrcPad. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1137>
2023-03-15 13:59:50 +00:00
"type": "GstTranslateSrcPad"
net/aws/transcriber: translate: optional experimental translation tokenization This commit adds an optional experimental translation tokenization feature. It can be activated using the `translation_src_%u` pads property `tokenization-method`. For the moment, the feature is deactivated by default. The Translate ws accepts '<span></span>' tags in the input and adds matching tags in the output. When an 'id' is also provided as an attribute of the 'span', the matching output tag also uses this 'id'. In the context of close captions, the 'id's are of little use. However, we can take advantage of the spans in the output to identify translation chunks, which more or less reflect the rythm of the input transcript. This commit adds simples spans (no 'id') to the input Transcript Items and parses the resulting spans in the translated output, assigning the timestamps and durations sequentially from the input Transcript Items. Edge cases such as absence of spans, nested spans were observed and are handled here. Similarly, mismatches between the number of input and output items are taken care of by some sort of reconcialiation. Note that this is still experimental and requires further testings. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1109>
2023-03-11 15:27:51 +00:00
},
net/aws/transcriber: rename prop transcript-lookahead & TranslationSrcPad ... as translate-lookahead and TranslateSrcPad. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1137>
2023-03-15 13:59:50 +00:00
"translate_src_%%u": {
net/aws/transcriber: translate: optional experimental translation tokenization This commit adds an optional experimental translation tokenization feature. It can be activated using the `translation_src_%u` pads property `tokenization-method`. For the moment, the feature is deactivated by default. The Translate ws accepts '<span></span>' tags in the input and adds matching tags in the output. When an 'id' is also provided as an attribute of the 'span', the matching output tag also uses this 'id'. In the context of close captions, the 'id's are of little use. However, we can take advantage of the spans in the output to identify translation chunks, which more or less reflect the rythm of the input transcript. This commit adds simples spans (no 'id') to the input Transcript Items and parses the resulting spans in the translated output, assigning the timestamps and durations sequentially from the input Transcript Items. Edge cases such as absence of spans, nested spans were observed and are handled here. Similarly, mismatches between the number of input and output items are taken care of by some sort of reconcialiation. Note that this is still experimental and requires further testings. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1109>
2023-03-11 15:27:51 +00:00
"caps": "text/x-raw:\n format: utf8\n",
"direction": "src",
"presence": "request",
net/aws/transcriber: rename prop transcript-lookahead & TranslationSrcPad ... as translate-lookahead and TranslateSrcPad. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1137>
2023-03-15 13:59:50 +00:00
"type": "GstTranslateSrcPad"
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
}
},
"properties": {
"access-key": {
"blurb": "AWS Access Key",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
},
"language-code": {
"blurb": "The Language of the Stream, see <https://docs.aws.amazon.com/transcribe/latest/dg/how-streaming-transcription.html> for an up to date list of allowed languages",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "en-US",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
},
"latency": {
net/aws/transcriber: add translation request src pads This commit adds an optional transcript translation feature implemented as request src Pads. When requesting a src Pad, the user can specify the translation language code using Pad properties 'language-code'. The following properties are defined on the Element: - 'transcribe-latency': formerly 'latency', defines the expected latency for the Transcribe webservice. - 'translate-latency': defines the expected latency for the Translate webservice. - 'transcript-lookahead': maximum transcript duration to send to translation when a transcript is hitting its deadline and no punctuation was found. When the input and output languages are the same, only the 'transcribe-latency' is used for the Pad. Otherwise, the resulting latency is the addition of 'transcribe-latency' and 'translate-latency'. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1109>
2023-03-10 13:47:38 +00:00
"blurb": "Amount of milliseconds to allow AWS transcribe (Deprecated. Use transcribe-latency)",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "8000",
"max": "-1",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint",
"writable": true
},
"lateness": {
"blurb": "Amount of milliseconds to introduce as lateness",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "-1",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint",
"writable": true
},
"results-stability": {
"blurb": "Defines how fast results should stabilize",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "low (2)",
"mutable": "ready",
"readable": true,
"type": "GstAwsTranscriberResultStability",
"writable": true
},
"secret-access-key": {
"blurb": "AWS Secret Access Key",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
},
"session-id": {
"blurb": "The ID of the transcription session, must be length 36",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
},
"session-token": {
"blurb": "AWS temporary Session Token from STS",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
},
net/aws/transcriber: add translation request src pads This commit adds an optional transcript translation feature implemented as request src Pads. When requesting a src Pad, the user can specify the translation language code using Pad properties 'language-code'. The following properties are defined on the Element: - 'transcribe-latency': formerly 'latency', defines the expected latency for the Transcribe webservice. - 'translate-latency': defines the expected latency for the Translate webservice. - 'transcript-lookahead': maximum transcript duration to send to translation when a transcript is hitting its deadline and no punctuation was found. When the input and output languages are the same, only the 'transcribe-latency' is used for the Pad. Otherwise, the resulting latency is the addition of 'transcribe-latency' and 'translate-latency'. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1109>
2023-03-10 13:47:38 +00:00
"transcribe-latency": {
"blurb": "Amount of milliseconds to allow AWS transcribe",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "8000",
"max": "-1",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint",
"writable": true
},
net/aws/transcriber: rename prop transcript-lookahead & TranslationSrcPad ... as translate-lookahead and TranslateSrcPad. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1137>
2023-03-15 13:59:50 +00:00
"translate-latency": {
"blurb": "Amount of milliseconds to allow AWS translate (ignored if the input and output languages are the same)",
net/aws/transcriber: add translation request src pads This commit adds an optional transcript translation feature implemented as request src Pads. When requesting a src Pad, the user can specify the translation language code using Pad properties 'language-code'. The following properties are defined on the Element: - 'transcribe-latency': formerly 'latency', defines the expected latency for the Transcribe webservice. - 'translate-latency': defines the expected latency for the Translate webservice. - 'transcript-lookahead': maximum transcript duration to send to translation when a transcript is hitting its deadline and no punctuation was found. When the input and output languages are the same, only the 'transcribe-latency' is used for the Pad. Otherwise, the resulting latency is the addition of 'transcribe-latency' and 'translate-latency'. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1109>
2023-03-10 13:47:38 +00:00
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
net/aws/transcriber: rename prop transcript-lookahead & TranslationSrcPad ... as translate-lookahead and TranslateSrcPad. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1137>
2023-03-15 13:59:50 +00:00
"default": "500",
net/aws/transcriber: add translation request src pads This commit adds an optional transcript translation feature implemented as request src Pads. When requesting a src Pad, the user can specify the translation language code using Pad properties 'language-code'. The following properties are defined on the Element: - 'transcribe-latency': formerly 'latency', defines the expected latency for the Transcribe webservice. - 'translate-latency': defines the expected latency for the Translate webservice. - 'transcript-lookahead': maximum transcript duration to send to translation when a transcript is hitting its deadline and no punctuation was found. When the input and output languages are the same, only the 'transcribe-latency' is used for the Pad. Otherwise, the resulting latency is the addition of 'transcribe-latency' and 'translate-latency'. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1109>
2023-03-10 13:47:38 +00:00
"max": "-1",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint",
"writable": true
},
net/aws/transcriber: rename prop transcript-lookahead & TranslationSrcPad ... as translate-lookahead and TranslateSrcPad. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1137>
2023-03-15 13:59:50 +00:00
"translate-lookahead": {
"blurb": "Maximum duration in milliseconds of transcript to lookahead before sending to translation when no separator was encountered",
net/aws/transcriber: add translation request src pads This commit adds an optional transcript translation feature implemented as request src Pads. When requesting a src Pad, the user can specify the translation language code using Pad properties 'language-code'. The following properties are defined on the Element: - 'transcribe-latency': formerly 'latency', defines the expected latency for the Transcribe webservice. - 'translate-latency': defines the expected latency for the Translate webservice. - 'transcript-lookahead': maximum transcript duration to send to translation when a transcript is hitting its deadline and no punctuation was found. When the input and output languages are the same, only the 'transcribe-latency' is used for the Pad. Otherwise, the resulting latency is the addition of 'transcribe-latency' and 'translate-latency'. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1109>
2023-03-10 13:47:38 +00:00
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
net/aws/transcriber: fix translate lookahead Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1137>
2023-03-16 09:29:24 +00:00
"default": "3000",
net/aws/transcriber: add translation request src pads This commit adds an optional transcript translation feature implemented as request src Pads. When requesting a src Pad, the user can specify the translation language code using Pad properties 'language-code'. The following properties are defined on the Element: - 'transcribe-latency': formerly 'latency', defines the expected latency for the Transcribe webservice. - 'translate-latency': defines the expected latency for the Translate webservice. - 'transcript-lookahead': maximum transcript duration to send to translation when a transcript is hitting its deadline and no punctuation was found. When the input and output languages are the same, only the 'transcribe-latency' is used for the Pad. Otherwise, the resulting latency is the addition of 'transcribe-latency' and 'translate-latency'. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1109>
2023-03-10 13:47:38 +00:00
"max": "-1",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint",
"writable": true
},
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2022-08-25 22:30:08 +00:00
"vocabulary-filter-method": {
"blurb": "Defines how filtered words will be edited, has no effect when vocabulary-filter-name isn't set",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "mask (0)",
"mutable": "ready",
"readable": true,
"type": "GstAwsTranscriberVocabularyFilterMethod",
"writable": true
},
"vocabulary-filter-name": {
"blurb": "The name of a custom filter vocabulary, see <https://docs.aws.amazon.com/transcribe/latest/help-panel/vocab-filter.html> for more information",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
},
"vocabulary-name": {
"blurb": "The name of a custom vocabulary, see <https://docs.aws.amazon.com/transcribe/latest/dg/how-vocabulary.html> for more information",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
}
},
"rank": "none"
}
},
"filename": "gstaws",
"license": "MPL",
"other-types": {
"GstAwsTranscriberResultStability": {
"kind": "enum",
"values": [
{
"desc": "High: stabilize results as fast as possible",
"name": "high",
"value": "0"
},
{
"desc": "Medium: balance between stability and accuracy",
"name": "medium",
"value": "1"
},
{
"desc": "Low: relatively less stable partial transcription results with higher accuracy",
"name": "low",
"value": "2"
}
]
},
net/aws/transcriber: translate: optional experimental translation tokenization This commit adds an optional experimental translation tokenization feature. It can be activated using the `translation_src_%u` pads property `tokenization-method`. For the moment, the feature is deactivated by default. The Translate ws accepts '<span></span>' tags in the input and adds matching tags in the output. When an 'id' is also provided as an attribute of the 'span', the matching output tag also uses this 'id'. In the context of close captions, the 'id's are of little use. However, we can take advantage of the spans in the output to identify translation chunks, which more or less reflect the rythm of the input transcript. This commit adds simples spans (no 'id') to the input Transcript Items and parses the resulting spans in the translated output, assigning the timestamps and durations sequentially from the input Transcript Items. Edge cases such as absence of spans, nested spans were observed and are handled here. Similarly, mismatches between the number of input and output items are taken care of by some sort of reconcialiation. Note that this is still experimental and requires further testings. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1109>
2023-03-11 15:27:51 +00:00
"GstAwsTranscriberTranslationTokenizationMethod": {
"kind": "enum",
"values": [
{
"desc": "None: don't tokenize translations",
"name": "none",
"value": "0"
},
{
"desc": "Span based: insert spans in the transript text and use the resulting spans in the translations to reproduce speech pacing.",
"name": "span-based",
"value": "1"
}
]
},
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2022-08-25 22:30:08 +00:00
"GstAwsTranscriberVocabularyFilterMethod": {
"kind": "enum",
"values": [
{
"desc": "Mask: replace words with ***",
"name": "mask",
"value": "0"
},
{
"desc": "Remove: delete words",
"name": "remove",
"value": "1"
},
{
"desc": "Tag: flag words without changing them",
"name": "tag",
"value": "2"
}
]
},
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>
2024-04-17 16:11:06 +00:00
"GstS3PutObjectSinkNextFile": {
"kind": "enum",
"values": [
{
"desc": "New file for each buffer",
"name": "next-buffer",
"value": "0"
},
{
"desc": "New file after each discontinuity",
"name": "next-discont",
"value": "1"
},
{
"desc": "New file at each key frame",
"name": "next-key-frame",
"value": "2"
},
{
"desc": "New file after a force key unit event",
"name": "next-key-unit-event",
"value": "3"
},
{
"desc": "New file when the configured maximum file size would be exceeded with the next buffer or buffer list",
"name": "next-max-size",
"value": "4"
},
{
"desc": "New file when the configured maximum duration would be exceeded with the next buffer or buffer list",
"name": "next-max-duration",
"value": "5"
}
]
},
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"GstS3SinkOnError": {
"kind": "enum",
"values": [
{
"desc": "Abort: Abort multipart upload on error.",
"name": "abort",
"value": "0"
},
{
"desc": "Complete: Complete multipart upload on error.",
"name": "complete",
"value": "1"
},
{
"desc": "DoNothing: Do nothing on error.",
"name": "nothing",
"value": "2"
}
]
net/aws/transcriber: add translation request src pads This commit adds an optional transcript translation feature implemented as request src Pads. When requesting a src Pad, the user can specify the translation language code using Pad properties 'language-code'. The following properties are defined on the Element: - 'transcribe-latency': formerly 'latency', defines the expected latency for the Transcribe webservice. - 'translate-latency': defines the expected latency for the Translate webservice. - 'transcript-lookahead': maximum transcript duration to send to translation when a transcript is hitting its deadline and no punctuation was found. When the input and output languages are the same, only the 'transcribe-latency' is used for the Pad. Otherwise, the resulting latency is the addition of 'transcribe-latency' and 'translate-latency'. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1109>
2023-03-10 13:47:38 +00:00
},
net/aws/transcriber: rename prop transcript-lookahead & TranslationSrcPad ... as translate-lookahead and TranslateSrcPad. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1137>
2023-03-15 13:59:50 +00:00
"GstTranslateSrcPad": {
net/aws/transcriber: add translation request src pads This commit adds an optional transcript translation feature implemented as request src Pads. When requesting a src Pad, the user can specify the translation language code using Pad properties 'language-code'. The following properties are defined on the Element: - 'transcribe-latency': formerly 'latency', defines the expected latency for the Transcribe webservice. - 'translate-latency': defines the expected latency for the Translate webservice. - 'transcript-lookahead': maximum transcript duration to send to translation when a transcript is hitting its deadline and no punctuation was found. When the input and output languages are the same, only the 'transcribe-latency' is used for the Pad. Otherwise, the resulting latency is the addition of 'transcribe-latency' and 'translate-latency'. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1109>
2023-03-10 13:47:38 +00:00
"hierarchy": [
net/aws/transcriber: rename prop transcript-lookahead & TranslationSrcPad ... as translate-lookahead and TranslateSrcPad. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1137>
2023-03-15 13:59:50 +00:00
"GstTranslateSrcPad",
net/aws/transcriber: add translation request src pads This commit adds an optional transcript translation feature implemented as request src Pads. When requesting a src Pad, the user can specify the translation language code using Pad properties 'language-code'. The following properties are defined on the Element: - 'transcribe-latency': formerly 'latency', defines the expected latency for the Transcribe webservice. - 'translate-latency': defines the expected latency for the Translate webservice. - 'transcript-lookahead': maximum transcript duration to send to translation when a transcript is hitting its deadline and no punctuation was found. When the input and output languages are the same, only the 'transcribe-latency' is used for the Pad. Otherwise, the resulting latency is the addition of 'transcribe-latency' and 'translate-latency'. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1109>
2023-03-10 13:47:38 +00:00
"GstPad",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"kind": "object",
"properties": {
"language-code": {
"blurb": "The Language the Stream must be translated to",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
net/aws/transcriber: translate: optional experimental translation tokenization This commit adds an optional experimental translation tokenization feature. It can be activated using the `translation_src_%u` pads property `tokenization-method`. For the moment, the feature is deactivated by default. The Translate ws accepts '<span></span>' tags in the input and adds matching tags in the output. When an 'id' is also provided as an attribute of the 'span', the matching output tag also uses this 'id'. In the context of close captions, the 'id's are of little use. However, we can take advantage of the spans in the output to identify translation chunks, which more or less reflect the rythm of the input transcript. This commit adds simples spans (no 'id') to the input Transcript Items and parses the resulting spans in the translated output, assigning the timestamps and durations sequentially from the input Transcript Items. Edge cases such as absence of spans, nested spans were observed and are handled here. Similarly, mismatches between the number of input and output items are taken care of by some sort of reconcialiation. Note that this is still experimental and requires further testings. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1109>
2023-03-11 15:27:51 +00:00
},
"tokenization-method": {
"blurb": "The tokenization method to apply to translations",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "none (0)",
"mutable": "ready",
"readable": true,
"type": "GstAwsTranscriberTranslationTokenizationMethod",
"writable": true
net/aws/transcriber: add translation request src pads This commit adds an optional transcript translation feature implemented as request src Pads. When requesting a src Pad, the user can specify the translation language code using Pad properties 'language-code'. The following properties are defined on the Element: - 'transcribe-latency': formerly 'latency', defines the expected latency for the Transcribe webservice. - 'translate-latency': defines the expected latency for the Translate webservice. - 'transcript-lookahead': maximum transcript duration to send to translation when a transcript is hitting its deadline and no punctuation was found. When the input and output languages are the same, only the 'transcribe-latency' is used for the Pad. Otherwise, the resulting latency is the addition of 'transcribe-latency' and 'translate-latency'. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1109>
2023-03-10 13:47:38 +00:00
}
}
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
}
},
"package": "gst-plugin-aws",
"source": "gst-plugin-aws",
"tracers": {},
"url": "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
},
"cdg": {
Update docs
2022-10-23 17:35:14 +00:00
"description": "GStreamer CDG codec Plugin",
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2022-08-25 22:30:08 +00:00
"elements": {
"cdgdec": {
"author": "Guillaume Desmottes <guillaume.desmottes@collabora.com>",
"description": "CDG decoder",
"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstCdgDec",
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2022-08-25 22:30:08 +00:00
"GstVideoDecoder",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Decoder/Video",
"long-name": "CDG decoder",
"pad-templates": {
"sink": {
"caps": "video/x-cdg:\n parsed: true\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "video/x-raw:\n format: RGBA\n width: 300\n height: 216\n framerate: 0/1\n",
"direction": "src",
"presence": "always"
}
},
"rank": "primary"
},
"cdgparse": {
"author": "Guillaume Desmottes <guillaume.desmottes@collabora.com>",
"description": "CDG parser",
"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstCdgParse",
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2022-08-25 22:30:08 +00:00
"GstBaseParse",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Codec/Parser/Video",
"long-name": "CDG parser",
"pad-templates": {
"sink": {
"caps": "video/x-cdg:\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "video/x-cdg:\n width: 300\n height: 216\n framerate: 0/1\n parsed: true\n",
"direction": "src",
"presence": "always"
}
},
"rank": "primary"
}
},
"filename": "gstcdg",
"license": "MIT/X11",
"other-types": {},
"package": "gst-plugin-cdg",
"source": "gst-plugin-cdg",
"tracers": {},
"url": "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
},
"claxon": {
Update docs
2022-10-23 17:35:14 +00:00
"description": "GStreamer Claxon FLAC Decoder Plugin",
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2022-08-25 22:30:08 +00:00
"elements": {
"claxondec": {
"author": "Ruben Gonzalez <rgonzalez@fluendo.com>",
"description": "Claxon FLAC decoder",
"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstClaxonDec",
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2022-08-25 22:30:08 +00:00
"GstAudioDecoder",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Decoder/Audio",
"long-name": "Claxon FLAC decoder",
"pad-templates": {
"sink": {
"caps": "audio/x-flac:\n framed: true\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "audio/x-raw:\n rate: [ 1, 655349 ]\n channels: [ 1, 7 ]\n layout: interleaved\n format: { S8, S16LE, S24_32LE, S32LE }\n",
"direction": "src",
"presence": "always"
}
},
"rank": "marginal"
}
},
"filename": "gstclaxon",
"license": "MIT/X11",
"other-types": {},
"package": "gst-plugin-claxon",
"source": "gst-plugin-claxon",
"tracers": {},
"url": "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
},
"csound": {
Update docs
2022-10-23 17:35:14 +00:00
"description": "GStreamer Audio Filter plugin based on Csound",
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2022-08-25 22:30:08 +00:00
"elements": {
"csoundfilter": {
"author": "Natanael Mojica <neithanmo@gmail.com>",
"description": "Implement an audio filter/effects using Csound",
"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstCsoundFilter",
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2022-08-25 22:30:08 +00:00
"GstBaseTransform",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Filter/Effect/Audio",
"long-name": "Audio filter",
"pad-templates": {
"sink": {
"caps": "audio/x-raw:\n rate: [ 1, 2147483647 ]\n channels: [ 1, 2147483647 ]\n layout: interleaved\n format: F64LE\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "audio/x-raw:\n rate: [ 1, 2147483647 ]\n channels: [ 1, 2147483647 ]\n layout: interleaved\n format: F64LE\n",
"direction": "src",
"presence": "always"
}
},
"properties": {
"csd-text": {
"blurb": "The content of a csd file passed as a String.\n Use either location or csd-text but not both at the same time, if so and error would be triggered",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
},
"location": {
"blurb": "Location of the csd file to be used by csound.\n Use either location or CSD-text but not both at the same time, if so and error would be triggered",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
},
"loop": {
"blurb": "loop over the score (can be changed in PLAYING or PAUSED state)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "playing",
"readable": true,
"type": "gboolean",
"writable": true
},
"score-offset": {
"blurb": "Score offset in seconds to start the performance",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "1.79769e+308",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "gdouble",
"writable": true
}
},
"rank": "none"
}
},
"filename": "gstcsound",
"license": "MIT/X11",
"other-types": {},
"package": "gst-plugin-csound",
"source": "gst-plugin-csound",
"tracers": {},
"url": "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
},
Update docs
2022-10-23 17:35:14 +00:00
"dav1d": {
"description": "GStreamer dav1d AV1 decoder Plugin",
"elements": {
"dav1ddec": {
"author": "Philippe Normand <philn@igalia.com>",
"description": "Decode AV1 video streams with dav1d",
"hierarchy": [
"GstDav1dDec",
"GstVideoDecoder",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Codec/Decoder/Video",
"long-name": "Dav1d AV1 Decoder",
"pad-templates": {
"sink": {
"caps": "video/x-av1:\n stream-format: obu-stream\n alignment: { (string)frame, (string)tu }\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "video/x-raw:\n format: { GRAY8, GRAY16_LE, I420, Y42B, Y444, I420_10LE, I422_10LE, Y444_10LE, I420_12LE, I422_12LE, Y444_12LE }\n width: [ 1, 2147483647 ]\n height: [ 1, 2147483647 ]\n framerate: [ 0/1, 2147483647/1 ]\n",
"direction": "src",
"presence": "always"
}
},
"properties": {
"max-frame-delay": {
"blurb": "Maximum delay in frames for the decoder (set to 1 for low latency, 0 to be equal to the number of logical cores. -1 to choose between these two based on pipeline liveness)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "18446744073709551615",
"max": "4294967295",
"min": "-1",
"mutable": "ready",
"readable": true,
"type": "gint64",
"writable": true
},
"n-threads": {
"blurb": "Number of threads to use while decoding (set to 0 to use number of logical cores)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "-1",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint",
"writable": true
}
},
dav1ddec: Lower rank to primary The rank of AOM av1dec was demoted as secondary, and thus primary rank is sufficient. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/996>
2022-11-30 18:35:39 +00:00
"rank": "primary"
Update docs
2022-10-23 17:35:14 +00:00
}
},
"filename": "gstdav1d",
"license": "MIT/X11",
"other-types": {},
"package": "gst-plugin-dav1d",
"source": "gst-plugin-dav1d",
"tracers": {},
"url": "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
},
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"fallbackswitch": {
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2022-10-23 17:35:14 +00:00
"description": "GStreamer Fallback Switcher and Source Plugin",
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2022-08-25 22:30:08 +00:00
"elements": {
"fallbacksrc": {
"author": "Sebastian Dröge <sebastian@centricular.com>",
fallbacksrc: Implement support for fallback streams
2022-08-19 14:34:17 +00:00
"description": "Live source with uridecodebin3 or custom source, and fallback stream",
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"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstFallbackSrc",
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2022-08-25 22:30:08 +00:00
"GstBin",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"interfaces": [
"GstChildProxy"
],
"klass": "Generic/Source",
"long-name": "Fallback Source",
"pad-templates": {
"audio": {
"caps": "ANY",
"direction": "src",
"presence": "sometimes"
},
"video": {
"caps": "ANY",
"direction": "src",
"presence": "sometimes"
}
},
"properties": {
"buffer-duration": {
"blurb": "Buffer duration when buffering streams (-1 default value)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "18446744073709551615",
"max": "9223372036854775806",
"min": "-1",
"mutable": "ready",
"readable": true,
"type": "gint64",
"writable": true
},
"enable-audio": {
"blurb": "Enable the audio stream, this will output silence if there's no audio in the configured URI",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "true",
"mutable": "ready",
"readable": true,
"type": "gboolean",
"writable": true
},
"enable-video": {
"blurb": "Enable the video stream, this will output black or the fallback video if there's no video in the configured URI",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "true",
"mutable": "ready",
"readable": true,
"type": "gboolean",
"writable": true
},
"fallback-audio-caps": {
"blurb": "Raw audio caps for fallback stream",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "ANY",
"mutable": "ready",
"readable": true,
"type": "GstCaps",
"writable": true
},
"fallback-uri": {
"blurb": "Fallback URI to use for video in case the main stream doesn't work",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
},
"fallback-video-caps": {
"blurb": "Raw video caps for fallback stream",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "ANY",
"mutable": "ready",
"readable": true,
"type": "GstCaps",
"writable": true
},
"immediate-fallback": {
"blurb": "Forward the fallback streams immediately at startup, when the primary streams are slow to start up and immediate output is required",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "ready",
"readable": true,
"type": "gboolean",
"writable": true
},
"manual-unblock": {
"blurb": "When enabled, the application must call the unblock signal, except for live streams",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "ready",
"readable": true,
"type": "gboolean",
"writable": true
},
"min-latency": {
"blurb": "When the main source has a higher latency than the fallback source this allows to configure a minimum latency that would be configured if initially the fallback is enabled",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "18446744073709551614",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint64",
"writable": true
},
"restart-on-eos": {
"blurb": "Restart source on EOS",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "ready",
"readable": true,
"type": "gboolean",
"writable": true
},
"restart-timeout": {
"blurb": "Timeout for restarting an active source",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "5000000000",
"max": "18446744073709551614",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint64",
"writable": true
},
"retry-timeout": {
"blurb": "Timeout for stopping after repeated failure",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "60000000000",
"max": "18446744073709551614",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint64",
"writable": true
},
"source": {
"blurb": "Source to use instead of the URI",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"mutable": "ready",
"readable": true,
"type": "GstElement",
"writable": true
},
"statistics": {
"blurb": "Various statistics",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
fallbacksrc: Implement support for fallback streams
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"default": "application/x-fallbacksrc-stats, num-retry=(guint64)0, num-fallback-retry=(guint64)0, last-retry-reason=(GstFallbackSourceRetryReason)none, last-fallback-retry-reason=(GstFallbackSourceRetryReason)none, buffering-percent=(int)100, fallback-buffering-percent=(int)100;",
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"mutable": "null",
"readable": true,
"type": "GstStructure",
"writable": false
},
"status": {
"blurb": "Current source status",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "stopped (0)",
"mutable": "null",
"readable": true,
"type": "GstFallbackSourceStatus",
"writable": false
},
"timeout": {
"blurb": "Timeout for switching to the fallback URI",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "5000000000",
"max": "18446744073709551614",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint64",
"writable": true
},
"uri": {
"blurb": "URI to use",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
}
},
"rank": "none",
"signals": {
"unblock": {
"action": true,
"args": [],
"return-type": "void",
"when": "last"
},
"update-uri": {
"args": [
{
"name": "arg0",
"type": "gchararray"
}
],
"return-type": "gchararray",
"when": "last"
}
}
},
"fallbackswitch": {
"author": "Jan Schmidt <jan@centricular.com>",
"description": "Priority-based automatic input selector element",
"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstFallbackSwitch",
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2022-08-25 22:30:08 +00:00
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"interfaces": [
"GstChildProxy"
],
"klass": "Generic",
"long-name": "Priority-based input selector",
"pad-templates": {
"sink_%%u": {
"caps": "ANY",
"direction": "sink",
"presence": "request",
Update docs
2022-10-23 17:35:14 +00:00
"type": "GstFallbackSwitchSinkPad"
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},
"src": {
"caps": "ANY",
"direction": "src",
"presence": "always"
}
},
"properties": {
"active-pad": {
"blurb": "Currently active pad",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"mutable": "playing",
"readable": true,
"type": "GstPad",
"writable": true
},
"auto-switch": {
"blurb": "Automatically switch pads (If true, use the priority pad property, otherwise manual selection via the active-pad property)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "true",
"mutable": "ready",
"readable": true,
"type": "gboolean",
"writable": true
},
"immediate-fallback": {
"blurb": "Forward lower-priority streams immediately at startup, when the stream with priority 0 is slow to start up and immediate output is required",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "ready",
"readable": true,
"type": "gboolean",
"writable": true
},
"latency": {
"blurb": "Additional latency in live mode to allow upstream to take longer to produce buffers for the current position (in nanoseconds)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "18446744073709551614",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint64",
"writable": true
},
"min-upstream-latency": {
"blurb": "When sources with a higher latency are expected to be plugged in dynamically after the fallbackswitch has started playing, this allows overriding the minimum latency reported by the initial source(s). This is only taken into account when larger than the actually reported minimum latency. (nanoseconds)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "18446744073709551614",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint64",
"writable": true
},
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>
2023-06-12 15:48:54 +00:00
"stop-on-eos": {
"blurb": "Stop forwarding buffers as soon as one input pad is eos",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "ready",
"readable": true,
"type": "gboolean",
"writable": true
},
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"timeout": {
"blurb": "Timeout on an input before switching to a lower priority input.",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "1000000000",
"max": "18446744073709551614",
"min": "0",
"mutable": "playing",
"readable": true,
"type": "guint64",
"writable": true
}
},
"rank": "none"
}
},
"filename": "gstfallbackswitch",
"license": "MPL",
"other-types": {
Update docs
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"GstFallbackSourceStatus": {
"kind": "enum",
"values": [
{
"desc": "Stopped",
"name": "stopped",
"value": "0"
},
{
"desc": "Buffering",
"name": "buffering",
"value": "1"
},
{
"desc": "Retrying",
"name": "retrying",
"value": "2"
},
{
"desc": "Running",
"name": "running",
"value": "3"
}
]
},
"GstFallbackSwitchSinkPad": {
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"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstFallbackSwitchSinkPad",
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2022-08-25 22:30:08 +00:00
"GstPad",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"kind": "object",
"properties": {
"is-healthy": {
"blurb": "Whether this stream is healthy",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "null",
"readable": true,
"type": "gboolean",
"writable": false
},
"priority": {
fallbackswitch: document the pad priority ordering I just wasted lots of time trying to figure out why my higher priority pad wasn't used... Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1208>
2023-05-11 13:20:41 +00:00
"blurb": "Selection priority for this stream (lower number has a higher priority)",
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"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "-1",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
}
}
}
},
"package": "gst-plugin-fallbackswitch",
"source": "gst-plugin-fallbackswitch",
"tracers": {},
"url": "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
},
"ffv1": {
Update docs
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"description": "GStreamer FFV1 Decoder Plugin",
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"elements": {
"ffv1dec": {
"author": "Arun Raghavan <arun@asymptotic.io>",
"description": "Decode FFV1 video streams",
"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstFfv1Dec",
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2022-08-25 22:30:08 +00:00
"GstVideoDecoder",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Codec/Decoder/Video",
"long-name": "FFV1 Decoder",
"pad-templates": {
"sink": {
"caps": "video/x-ffv:\n ffvversion: 1\n width: [ 1, 2147483647 ]\n height: [ 1, 2147483647 ]\n framerate: [ 0/1, 2147483647/1 ]\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "video/x-raw:\n format: { GRAY8, GRAY16_LE, GRAY16_BE, Y444, Y444_10LE, Y444_10BE, A444_10LE, A444_10BE, Y444_12LE, Y444_12BE, Y444_16LE, Y444_16BE, A420, Y42B, I422_10LE, I422_10BE, A422_10LE, A422_10BE, I422_12LE, I422_12BE, I420, I420_10LE, I420_10BE, I420_12LE, I420_12BE, GBRA, GBR, GBR_10LE, GBR_10BE, GBRA_10LE, GBRA_10BE, GBR_12LE, GBR_12BE, GBRA_12LE, GBRA_12BE, Y41B, YUV9 }\n width: [ 1, 2147483647 ]\n height: [ 1, 2147483647 ]\n framerate: [ 0/1, 2147483647/1 ]\n",
"direction": "src",
"presence": "always"
}
},
ffv1dec: Drop rank for now We'll keep the rank lower than avdec_ffv1, at least until we're comfortable with support for the entire range of possible inputs working well. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1174>
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"rank": "none"
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}
},
"filename": "gstffv1",
"license": "MIT/X11",
"other-types": {},
"package": "gst-plugin-ffv1",
"source": "gst-plugin-ffv1",
"tracers": {},
"url": "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
},
"fmp4": {
Update docs
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"description": "GStreamer Fragmented MP4 Plugin",
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"elements": {
"cmafmux": {
"author": "Sebastian Dröge <sebastian@centricular.com>",
"description": "CMAF fragmented MP4 muxer",
"hierarchy": [
"GstCMAFMux",
"GstFMP4Mux",
"GstAggregator",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Codec/Muxer",
"long-name": "CMAFMux",
"pad-templates": {
"sink": {
cmafmux: Add opus support Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1692>
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"caps": "video/x-h264:\n stream-format: { (string)avc, (string)avc3 }\n alignment: au\n width: [ 1, 65535 ]\n height: [ 1, 65535 ]\nvideo/x-av1:\n stream-format: obu-stream\n alignment: tu\n profile: { (string)main, (string)high, (string)professional }\n chroma-format: { (string)4:0:0, (string)4:2:0, (string)4:2:2, (string)4:4:4 }\n bit-depth-luma: { (uint)8, (uint)10, (uint)12 }\nbit-depth-chroma: { (uint)8, (uint)10, (uint)12 }\n width: [ 1, 65535 ]\n height: [ 1, 65535 ]\nvideo/x-h265:\n stream-format: { (string)hvc1, (string)hev1 }\n alignment: au\n width: [ 1, 65535 ]\n height: [ 1, 65535 ]\naudio/mpeg:\n mpegversion: 4\n stream-format: raw\n channels: [ 1, 65535 ]\n rate: [ 1, 2147483647 ]\naudio/x-opus:\nchannel-mapping-family: [ 0, 255 ]\n channels: [ 1, 8 ]\n rate: [ 1, 2147483647 ]\n",
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"direction": "sink",
fmp4mux: Make media/trak timescales configurable And refactor a bit of code for easier extensibility.
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"presence": "always",
"type": "GstFMP4MuxPad"
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},
"src": {
"caps": "video/quicktime:\n variant: cmaf\n",
"direction": "src",
"presence": "always"
}
},
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>
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"rank": "primary",
"signals": {
"send-headers": {
"action": true,
"args": [],
"return-type": "void",
"when": "last"
}
}
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},
"dashmp4mux": {
"author": "Sebastian Dröge <sebastian@centricular.com>",
"description": "DASH fragmented MP4 muxer",
"hierarchy": [
"GstDASHMP4Mux",
"GstFMP4Mux",
"GstAggregator",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Codec/Muxer",
"long-name": "DASHMP4Mux",
"pad-templates": {
"sink": {
mp4/fmp4: Update docs Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1075>
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"caps": "video/x-h264:\n stream-format: { (string)avc, (string)avc3 }\n alignment: au\n width: [ 1, 65535 ]\n height: [ 1, 65535 ]\nvideo/x-h265:\n stream-format: { (string)hvc1, (string)hev1 }\n alignment: au\n width: [ 1, 65535 ]\n height: [ 1, 65535 ]\nvideo/x-vp8:\n width: [ 1, 65535 ]\n height: [ 1, 65535 ]\nvideo/x-vp9:\n profile: { (string)0, (string)1, (string)2, (string)3 }\n chroma-format: { (string)4:2:0, (string)4:2:2, (string)4:4:4 }\n bit-depth-luma: { (uint)8, (uint)10, (uint)12 }\nbit-depth-chroma: { (uint)8, (uint)10, (uint)12 }\n width: [ 1, 65535 ]\n height: [ 1, 65535 ]\nvideo/x-av1:\n stream-format: obu-stream\n alignment: tu\n profile: { (string)main, (string)high, (string)professional }\n chroma-format: { (string)4:0:0, (string)4:2:0, (string)4:2:2, (string)4:4:4 }\n bit-depth-luma: { (uint)8, (uint)10, (uint)12 }\nbit-depth-chroma: { (uint)8, (uint)10, (uint)12 }\n width: [ 1, 65535 ]\n height: [ 1, 65535 ]\naudio/mpeg:\n mpegversion: 4\n stream-format: raw\n channels: [ 1, 65535 ]\n rate: [ 1, 2147483647 ]\naudio/x-opus:\nchannel-mapping-family: [ 0, 255 ]\n channels: [ 1, 8 ]\n rate: [ 1, 2147483647 ]\n",
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"direction": "sink",
fmp4mux: Make media/trak timescales configurable And refactor a bit of code for easier extensibility.
2022-11-07 17:47:31 +00:00
"presence": "always",
"type": "GstFMP4MuxPad"
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},
"src": {
"caps": "video/quicktime:\n variant: iso-fragmented\n",
"direction": "src",
"presence": "always"
}
},
"rank": "primary"
},
"isofmp4mux": {
"author": "Sebastian Dröge <sebastian@centricular.com>",
"description": "ISO fragmented MP4 muxer",
"hierarchy": [
"GstISOFMP4Mux",
"GstFMP4Mux",
"GstAggregator",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Codec/Muxer",
"long-name": "ISOFMP4Mux",
"pad-templates": {
"sink_%%u": {
mp4/fmp4: support flac inside the iso (f)mp4 container Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1401>
2023-11-28 17:53:10 +00:00
"caps": "video/x-h264:\n stream-format: { (string)avc, (string)avc3 }\n alignment: au\n width: [ 1, 65535 ]\n height: [ 1, 65535 ]\nvideo/x-h265:\n stream-format: { (string)hvc1, (string)hev1 }\n alignment: au\n width: [ 1, 65535 ]\n height: [ 1, 65535 ]\nvideo/x-vp8:\n width: [ 1, 65535 ]\n height: [ 1, 65535 ]\nvideo/x-vp9:\n profile: { (string)0, (string)1, (string)2, (string)3 }\n chroma-format: { (string)4:2:0, (string)4:2:2, (string)4:4:4 }\n bit-depth-luma: { (uint)8, (uint)10, (uint)12 }\nbit-depth-chroma: { (uint)8, (uint)10, (uint)12 }\n width: [ 1, 65535 ]\n height: [ 1, 65535 ]\nvideo/x-av1:\n stream-format: obu-stream\n alignment: tu\n profile: { (string)main, (string)high, (string)professional }\n chroma-format: { (string)4:0:0, (string)4:2:0, (string)4:2:2, (string)4:4:4 }\n bit-depth-luma: { (uint)8, (uint)10, (uint)12 }\nbit-depth-chroma: { (uint)8, (uint)10, (uint)12 }\n width: [ 1, 65535 ]\n height: [ 1, 65535 ]\naudio/mpeg:\n mpegversion: 4\n stream-format: raw\n channels: [ 1, 65535 ]\n rate: [ 1, 2147483647 ]\naudio/x-opus:\nchannel-mapping-family: [ 0, 255 ]\n channels: [ 1, 8 ]\n rate: [ 1, 2147483647 ]\naudio/x-flac:\n framed: true\n channels: [ 1, 8 ]\n rate: [ 1, 655350 ]\n",
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"direction": "sink",
fmp4mux: Make media/trak timescales configurable And refactor a bit of code for easier extensibility.
2022-11-07 17:47:31 +00:00
"presence": "request",
"type": "GstFMP4MuxPad"
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},
"src": {
"caps": "video/quicktime:\n variant: iso-fragmented\n",
"direction": "src",
"presence": "always"
}
},
fmp4mux: add 'offset-to-zero' property Add it only to 'isofmp4mux', the onvif variant already does this and CMAF and DASH are always single-stream so you rely on inter-container synchronization via the running-time. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1063>
2023-01-25 09:59:52 +00:00
"properties": {
"offset-to-zero": {
"blurb": "Offsets all streams so that the earliest stream starts at 0",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "ready",
"readable": true,
"type": "gboolean",
"writable": true
}
},
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2022-08-25 22:30:08 +00:00
"rank": "primary"
},
"onviffmp4mux": {
"author": "Sebastian Dröge <sebastian@centricular.com>",
"description": "ONVIF fragmented MP4 muxer",
"hierarchy": [
"GstONVIFFMP4Mux",
"GstFMP4Mux",
"GstAggregator",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Codec/Muxer",
"long-name": "ONVIFFMP4Mux",
"pad-templates": {
"sink_%%u": {
"caps": "video/x-h264:\n stream-format: { (string)avc, (string)avc3 }\n alignment: au\n width: [ 1, 65535 ]\n height: [ 1, 65535 ]\nvideo/x-h265:\n stream-format: { (string)hvc1, (string)hev1 }\n alignment: au\n width: [ 1, 65535 ]\n height: [ 1, 65535 ]\nimage/jpeg:\n width: [ 1, 65535 ]\n height: [ 1, 65535 ]\naudio/mpeg:\n mpegversion: 4\n stream-format: raw\n channels: [ 1, 65535 ]\n rate: [ 1, 2147483647 ]\naudio/x-alaw:\n channels: [ 1, 2 ]\n rate: [ 1, 2147483647 ]\naudio/x-mulaw:\n channels: [ 1, 2 ]\n rate: [ 1, 2147483647 ]\naudio/x-adpcm:\n layout: g726\n channels: 1\n rate: 8000\n bitrate: { (int)16000, (int)24000, (int)32000, (int)40000 }\napplication/x-onvif-metadata:\n parsed: true\n",
"direction": "sink",
fmp4mux: Make media/trak timescales configurable And refactor a bit of code for easier extensibility.
2022-11-07 17:47:31 +00:00
"presence": "request",
"type": "GstFMP4MuxPad"
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
},
"src": {
"caps": "video/quicktime:\n variant: iso-fragmented\n",
"direction": "src",
"presence": "always"
}
},
"rank": "primary"
}
},
"filename": "gstfmp4",
"license": "MPL",
"other-types": {
"GstFMP4Mux": {
"hierarchy": [
"GstFMP4Mux",
"GstAggregator",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"kind": "object",
"properties": {
fmp4mux: Add support for sub-fragments / chunking Allow outputting sub-fragments (chunks in CMAF terms) that are shorter than the fragment duration and don't usually start on a keyframe. By this the buffering requirements of the element is reduced to one chunk duration, as is the latency. This is used for formats like low-latency / LL-HLS and DASH. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1059>
2023-01-23 18:43:26 +00:00
"chunk-duration": {
"blurb": "Duration for each FMP4 chunk (default = no chunks)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "18446744073709551615",
"max": "18446744073709551615",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint64",
"writable": true
},
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"fragment-duration": {
fmp4mux: specify the fragment duration unit The fragment duration is expressed in nanoseconds. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1348>
2023-10-04 08:17:47 +00:00
"blurb": "Duration for each FMP4 fragment in nanoseconds",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "10000000000",
"max": "18446744073709551615",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint64",
"writable": true
},
"header-update-mode": {
"blurb": "Mode for updating the header at the end of the stream",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "none (0)",
"mutable": "ready",
"readable": true,
"type": "GstFMP4MuxHeaderUpdateMode",
"writable": true
},
"interleave-bytes": {
"blurb": "Interleave between streams in bytes",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "18446744073709551615",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint64",
"writable": true
},
"interleave-time": {
"blurb": "Interleave between streams in nanoseconds",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "250000000",
"max": "18446744073709551615",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint64",
"writable": true
},
fmp4mux: Make media/trak timescales configurable And refactor a bit of code for easier extensibility.
2022-11-07 17:47:31 +00:00
"movie-timescale": {
"blurb": "Timescale to use for the movie (units per second, 0 is automatic)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "-1",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint",
"writable": true
},
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"write-mehd": {
"blurb": "Write movie extends header box with the duration at the end of the stream (needs a header-update-mode enabled)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "ready",
"readable": true,
"type": "gboolean",
"writable": true
},
"write-mfra": {
"blurb": "Write fragment random access box at the end of the stream",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "ready",
"readable": true,
"type": "gboolean",
"writable": true
}
}
},
"GstFMP4MuxHeaderUpdateMode": {
"kind": "enum",
"values": [
{
"desc": "None",
"name": "none",
"value": "0"
},
{
"desc": "Rewrite",
"name": "rewrite",
"value": "1"
},
{
"desc": "Update",
"name": "update",
"value": "2"
}
]
fmp4mux: Make media/trak timescales configurable And refactor a bit of code for easier extensibility.
2022-11-07 17:47:31 +00:00
},
"GstFMP4MuxPad": {
"hierarchy": [
"GstFMP4MuxPad",
"GstAggregatorPad",
"GstPad",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"kind": "object",
"properties": {
"trak-timescale": {
"blurb": "Timescale to use for the track (units per second, 0 is automatic)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "-1",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint",
"writable": true
}
}
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2022-08-25 22:30:08 +00:00
}
},
"package": "gst-plugin-fmp4",
"source": "gst-plugin-fmp4",
"tracers": {},
"url": "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
},
"gif": {
"description": "GStreamer GIF plugin",
"elements": {
"gifenc": {
"author": "Markus Ebner <info@ebner-markus.de>",
"description": "GIF encoder",
"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstGifEnc",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"GstVideoEncoder",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"interfaces": [
"GstPreset"
],
"klass": "Encoder/Video",
"long-name": "GIF encoder",
"pad-templates": {
"sink": {
"caps": "video/x-raw:\n format: { RGB, RGBA }\n width: [ 1, 2147483647 ]\n height: [ 1, 2147483647 ]\n framerate: [ 1/1, 2147483599/21474836 ]\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "image/gif:\n",
"direction": "src",
"presence": "always"
}
},
"properties": {
"repeat": {
"blurb": "Repeat (-1 to loop forever, 0 .. n finite repetitions)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "65535",
"min": "-1",
"mutable": "ready",
"readable": true,
"type": "gint",
"writable": true
},
"speed": {
"blurb": "Speed (1 .. 30; higher value yields faster encoding)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "10",
"max": "30",
"min": "1",
"mutable": "ready",
"readable": true,
"type": "gint",
"writable": true
}
},
"rank": "primary"
}
},
"filename": "gstgif",
"license": "MIT/X11",
"other-types": {},
"package": "gst-plugin-gif",
"source": "gst-plugin-gif",
"tracers": {},
"url": "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
},
gopbuffer: implement element buffering of an entire GOP Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1349>
2023-02-02 00:48:12 +00:00
"gopbuffer": {
"description": "Store complete groups of pictures at a time",
"elements": {
"gopbuffer": {
"author": "Matthew Waters <matthew@centricular.com>",
"description": "GOP Buffer",
"hierarchy": [
"GstGopBuffer",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Video",
"pad-templates": {
"video_sink": {
"caps": "video/x-h264:\n stream-format: { (string)avc, (string)avc3 }\n alignment: au\nvideo/x-h265:\n stream-format: { (string)hvc1, (string)hev1 }\n alignment: au\nvideo/x-vp8:\nvideo/x-vp9:\nvideo/x-av1:\n stream-format: obu-stream\n alignment: tu\n",
"direction": "sink",
"presence": "always"
},
"video_src": {
"caps": "video/x-h264:\n stream-format: { (string)avc, (string)avc3 }\n alignment: au\nvideo/x-h265:\n stream-format: { (string)hvc1, (string)hev1 }\n alignment: au\nvideo/x-vp8:\nvideo/x-vp9:\nvideo/x-av1:\n stream-format: obu-stream\n alignment: tu\n",
"direction": "src",
"presence": "always"
}
},
"properties": {
"max-size-time": {
"blurb": "The maximum duration to store (0=disable)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "18446744073709551615",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint64",
"writable": true
},
"minimum-duration": {
"blurb": "The minimum duration to store",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "1000000000",
"max": "18446744073709551615",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint64",
"writable": true
}
},
"rank": "primary"
}
},
"filename": "gstgopbuffer",
"license": "MPL",
"other-types": {},
"package": "gst-plugin-gopbuffer",
"source": "gst-plugin-gopbuffer",
"tracers": {},
"url": "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
},
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"gtk4": {
gtk4: Fix description of the plugin A paintable is not a widget and that aspect does not belong in the short description anyway. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1563>
2024-05-07 14:56:22 +00:00
"description": "GStreamer GTK 4 sink element",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"elements": {
"gtk4paintablesink": {
"author": "Bilal Elmoussaoui <bil.elmoussaoui@gmail.com>, Jordan Petridis <jordan@centricular.com>, Sebastian Dröge <sebastian@centricular.com>",
"description": "A GTK 4 Paintable sink",
"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstGtk4PaintableSink",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"GstVideoSink",
"GstBaseSink",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
gtk4paintablesink: Implement child proxy interface This allows setting properties on the paintable from gst-launch-1.0. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1547>
2024-04-24 13:22:57 +00:00
"interfaces": [
"GstChildProxy"
],
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"klass": "Sink/Video",
"long-name": "GTK 4 Paintable Sink",
"pad-templates": {
"sink": {
gtk4: Implement support for directly importing dmabufs Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/441 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1547>
2024-04-20 20:19:20 +00:00
"caps": "video/x-raw(memory:GLMemory, meta:GstVideoOverlayComposition):\n format: { RGBA, RGB }\n width: [ 1, 2147483647 ]\n height: [ 1, 2147483647 ]\n framerate: [ 0/1, 2147483647/1 ]\n texture-target: 2D\n\nvideo/x-raw(memory:GLMemory):\n format: { RGBA, RGB }\n width: [ 1, 2147483647 ]\n height: [ 1, 2147483647 ]\n framerate: [ 0/1, 2147483647/1 ]\n texture-target: 2D\n\nvideo/x-raw(memory:SystemMemory, meta:GstVideoOverlayComposition):\n format: { BGRA, ARGB, RGBA, ABGR, RGB, BGR }\n width: [ 1, 2147483647 ]\n height: [ 1, 2147483647 ]\n framerate: [ 0/1, 2147483647/1 ]\n\nvideo/x-raw(meta:GstVideoOverlayComposition):\n format: { BGRA, ARGB, RGBA, ABGR, RGB, BGR }\n width: [ 1, 2147483647 ]\n height: [ 1, 2147483647 ]\n framerate: [ 0/1, 2147483647/1 ]\nvideo/x-raw:\n format: { BGRA, ARGB, RGBA, ABGR, RGB, BGR }\n width: [ 1, 2147483647 ]\n height: [ 1, 2147483647 ]\n framerate: [ 0/1, 2147483647/1 ]\n",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"direction": "sink",
"presence": "always"
}
},
"properties": {
"paintable": {
"blurb": "The Paintable the sink renders to",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"mutable": "null",
"readable": true,
gtk4: Document paintable properties correctly Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1655>
2024-07-06 10:56:24 +00:00
"type": "GstGtk4Paintable",
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2022-08-25 22:30:08 +00:00
"writable": false
gtk4: add window-{width,height} property Allow the application to pass the actual rendering size so overlays can be rendered accordingly. Fix #562 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1680>
2024-07-25 11:28:14 +00:00
},
"window-height": {
"blurb": "the height of the main widget rendering the paintable",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "-1",
"min": "0",
"mutable": "playing",
"readable": true,
"type": "guint",
"writable": true
},
"window-width": {
"blurb": "the width of the main widget rendering the paintable",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "-1",
"min": "0",
"mutable": "playing",
"readable": true,
"type": "guint",
"writable": true
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
}
},
"rank": "none"
}
},
"filename": "gstgtk4",
"license": "MPL",
gtk4: Document paintable properties correctly Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1655>
2024-07-06 10:56:24 +00:00
"other-types": {
"GstGtk4Paintable": {
"hierarchy": [
"GstGtk4Paintable",
"GObject"
],
"interfaces": [
"GdkPaintable"
],
"kind": "object",
"properties": {
"background-color": {
"blurb": "Background color to render behind the video frame and in the borders",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "255",
"max": "-1",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
"force-aspect-ratio": {
"blurb": "When enabled, scaling will respect original aspect ratio",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "null",
"readable": true,
"type": "gboolean",
"writable": true
},
"gl-context": {
"blurb": "GL context to use for rendering",
"conditionally-available": false,
"construct": false,
"construct-only": true,
"controllable": false,
"mutable": "null",
"readable": true,
"type": "GdkGLContext",
"writable": true
},
gtk4: Add support for rotations / flipping Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/284 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1590>
2024-05-28 13:26:33 +00:00
"orientation": {
"blurb": "Orientation of the video frames",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "auto (0)",
"mutable": "null",
"readable": true,
"type": "GstGtk4PaintableSinkOrientation",
"writable": true
},
gtk4: Document paintable properties correctly Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1655>
2024-07-06 10:56:24 +00:00
"scaling-filter": {
"blurb": "Scaling filter to use for rendering",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "linear (0)",
"mutable": "null",
"readable": true,
"type": "GskScalingFilter",
"writable": true
},
"use-scaling-filter": {
"blurb": "Use selected scaling filter or GTK default for rendering",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "null",
"readable": true,
"type": "gboolean",
"writable": true
}
}
gtk4: Add support for rotations / flipping Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/284 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1590>
2024-05-28 13:26:33 +00:00
},
"GstGtk4PaintableSinkOrientation": {
"kind": "enum",
"values": [
{
"desc": "Auto",
"name": "auto",
"value": "0"
},
{
"desc": "Rotate0",
"name": "rotate0",
"value": "1"
},
{
"desc": "Rotate90",
"name": "rotate90",
"value": "2"
},
{
"desc": "Rotate180",
"name": "rotate180",
"value": "3"
},
{
"desc": "Rotate270",
"name": "rotate270",
"value": "4"
},
{
"desc": "FlipRotate0",
"name": "flip-rotate0",
"value": "5"
},
{
"desc": "FlipRotate90",
"name": "flip-rotate90",
"value": "6"
},
{
"desc": "FlipRotate180",
"name": "flip-rotate180",
"value": "7"
},
{
"desc": "FlipRotate270",
"name": "flip-rotate270",
"value": "8"
}
]
gtk4: Document paintable properties correctly Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1655>
2024-07-06 10:56:24 +00:00
}
},
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2022-08-25 22:30:08 +00:00
"package": "gst-plugin-gtk4",
"source": "gst-plugin-gtk4",
"tracers": {},
"url": "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
},
"hlssink3": {
Update docs
2022-10-23 17:35:14 +00:00
"description": "GStreamer HLS (HTTP Live Streaming) Plugin",
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2022-08-25 22:30:08 +00:00
"elements": {
hlssink3: Update plugin docs Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1306>
2023-09-24 11:07:46 +00:00
"hlscmafsink": {
"author": "Seungha Yang <seungha@centricular.com>",
"description": "HTTP Live Streaming CMAF Sink",
"hierarchy": [
"GstHlsCmafSink",
"GstHlsBaseSink",
"GstBin",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"interfaces": [
"GstChildProxy"
],
"klass": "Sink/Muxer",
"pad-templates": {
"sink": {
"caps": "video/x-h264:\n stream-format: { (string)avc, (string)avc3 }\n alignment: au\n width: [ 1, 65535 ]\n height: [ 1, 65535 ]\nvideo/x-h265:\n stream-format: { (string)hvc1, (string)hev1 }\n alignment: au\n width: [ 1, 65535 ]\n height: [ 1, 65535 ]\naudio/mpeg:\n mpegversion: 4\n stream-format: raw\n channels: [ 1, 65535 ]\n rate: [ 1, 2147483647 ]\n",
"direction": "sink",
"presence": "always"
}
},
"properties": {
"init-location": {
"blurb": "Location of the init fragment file to write",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "init%%05d.mp4",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
"latency": {
"blurb": "Additional latency to allow upstream to take longer to produce buffers for the current position (in nanoseconds)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "7500000000",
"max": "9223372036854775807",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint64",
"writable": true
},
"location": {
"blurb": "Location of the fragment file to write",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "segment%%05d.m4s",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
"playlist-type": {
"blurb": "The type of the playlist to use. When VOD type is set, the playlist will be live until the pipeline ends execution.",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "unspecified (0)",
"mutable": "ready",
"readable": true,
"type": "GstHlsSink3PlaylistType",
"writable": true
},
"sync": {
"blurb": "Sync on the clock",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "true",
"mutable": "null",
"readable": true,
"type": "gboolean",
"writable": true
},
"target-duration": {
"blurb": "The target duration in seconds of a segment/file. (0 - disabled, useful for management of segment duration by the streaming server)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "15",
"max": "-1",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint",
"writable": true
}
},
"rank": "none",
"signals": {
"get-init-stream": {
"args": [
{
"name": "arg0",
"type": "gchararray"
}
],
"return-type": "GOutputStream",
"when": "last"
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>
2024-08-19 07:58:41 +00:00
},
"new-playlist": {
"action": true,
"args": [],
"return-type": "void",
"when": "last"
hlssink3: Update plugin docs Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1306>
2023-09-24 11:07:46 +00:00
}
}
},
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"hlssink3": {
"author": "Alessandro Decina <alessandro.d@gmail.com>, Sebastian Dröge <sebastian@centricular.com>, Rafael Caricio <rafael@caricio.com>",
"description": "HTTP Live Streaming sink",
"hierarchy": [
"GstHlsSink3",
hlssink3: Update plugin docs Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1306>
2023-09-24 11:07:46 +00:00
"GstHlsBaseSink",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"GstBin",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"interfaces": [
"GstChildProxy"
],
"klass": "Sink/Muxer",
"pad-templates": {
"audio": {
"caps": "ANY",
"direction": "sink",
"presence": "request"
},
"video": {
"caps": "ANY",
"direction": "sink",
"presence": "request"
}
},
"properties": {
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>
2023-01-31 17:34:04 +00:00
"i-frames-only": {
"blurb": "Each video segments is single iframe, So put EXT-X-I-FRAMES-ONLY tag in the playlist",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "null",
"readable": true,
"type": "gboolean",
"writable": true
},
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"location": {
"blurb": "Location of the file to write",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "segment%%05d.ts",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
hlssink3: Update plugin docs Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1306>
2023-09-24 11:07:46 +00:00
"playlist-type": {
"blurb": "The type of the playlist to use. When VOD type is set, the playlist will be live until the pipeline ends execution.",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
hlssink3: Update plugin docs Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1306>
2023-09-24 11:07:46 +00:00
"default": "unspecified (0)",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"mutable": "null",
"readable": true,
hlssink3: Update plugin docs Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1306>
2023-09-24 11:07:46 +00:00
"type": "GstHlsSink3PlaylistType",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"writable": true
},
hlssink3: Update plugin docs Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1306>
2023-09-24 11:07:46 +00:00
"send-keyframe-requests": {
"blurb": "Send keyframe requests to ensure correct fragmentation. If this is disabled then the input must have keyframes in regular intervals.",
hlssink3: Add property track-pipeline-clock-for-pdt. This is required to take care of clock skew between system time and pipeline time. `track-pipeline-clock-for-pdt: true` mean utd time is sampled for first segment and for subsequent segments keep adding the time based on pipeline clock. difference of segment duration and PDT time will match. track-pipeline-clock-for-pdt: false` mean utd time is sampled for each segment. system time may jump forward or backward based on adjustments. If application needs to synchronization of external events `false` is recommended. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1145>
2023-04-25 07:41:43 +00:00
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "true",
"mutable": "null",
"readable": true,
"type": "gboolean",
"writable": true
},
hlssink3: Update plugin docs Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1306>
2023-09-24 11:07:46 +00:00
"target-duration": {
"blurb": "The target duration in seconds of a segment/file. (0 - disabled, useful for management of segment duration by the streaming server)",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
hlssink3: Update plugin docs Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1306>
2023-09-24 11:07:46 +00:00
"default": "15",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"max": "-1",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
hlssink3: Update plugin docs Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1306>
2023-09-24 11:07:46 +00:00
}
},
"rank": "none"
}
},
"filename": "gsthlssink3",
"license": "MPL",
"other-types": {
"GstHlsBaseSink": {
"hierarchy": [
"GstHlsBaseSink",
"GstBin",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"interfaces": [
"GstChildProxy"
],
"kind": "object",
"properties": {
"enable-endlist": {
"blurb": "Write \"EXT-X-ENDLIST\" tag to manifest at the end of stream",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
hlssink3: Update plugin docs Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1306>
2023-09-24 11:07:46 +00:00
"default": "true",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"mutable": "null",
"readable": true,
hlssink3: Update plugin docs Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1306>
2023-09-24 11:07:46 +00:00
"type": "gboolean",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"writable": true
},
hlssink3: Update plugin docs Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1306>
2023-09-24 11:07:46 +00:00
"enable-program-date-time": {
"blurb": "put EXT-X-PROGRAM-DATE-TIME tag in the playlist",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
hlssink3: Update plugin docs Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1306>
2023-09-24 11:07:46 +00:00
"default": "false",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"mutable": "null",
"readable": true,
hlssink3: Update plugin docs Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1306>
2023-09-24 11:07:46 +00:00
"type": "gboolean",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"writable": true
},
hlssink3: Update plugin docs Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1306>
2023-09-24 11:07:46 +00:00
"max-files": {
"blurb": "Maximum number of files to keep on disk. Once the maximum is reached, old files start to be deleted to make room for new ones.",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
hlssink3: Update plugin docs Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1306>
2023-09-24 11:07:46 +00:00
"default": "10",
"max": "-1",
"min": "0",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"mutable": "null",
"readable": true,
hlssink3: Update plugin docs Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1306>
2023-09-24 11:07:46 +00:00
"type": "guint",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"writable": true
},
hlssink3: Update plugin docs Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1306>
2023-09-24 11:07:46 +00:00
"pdt-follows-pipeline-clock": {
"blurb": "As there might be drift between the wallclock and pipeline clock, this controls whether the Program-Date-Time markers should follow the pipeline clock rate (true), or be skewed to match the wallclock rate (false).",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "true",
"mutable": "null",
"readable": true,
"type": "gboolean",
"writable": true
},
hlssink3: Update plugin docs Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1306>
2023-09-24 11:07:46 +00:00
"playlist-length": {
"blurb": "Length of HLS playlist. To allow players to conform to section 6.3.3 of the HLS specification, this should be at least 3. If set to 0, the playlist will be infinite.",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
hlssink3: Update plugin docs Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1306>
2023-09-24 11:07:46 +00:00
"default": "5",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"max": "-1",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
hlssink3: Update plugin docs Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1306>
2023-09-24 11:07:46 +00:00
},
"playlist-location": {
"blurb": "Location of the playlist to write.",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "playlist.m3u8",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
"playlist-root": {
"blurb": "Base path for the segments in the playlist file.",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
}
},
"signals": {
"delete-fragment": {
"args": [
{
"name": "arg0",
"type": "gchararray"
}
],
"return-type": "gboolean",
"when": "last"
},
"get-fragment-stream": {
"args": [
{
"name": "arg0",
"type": "gchararray"
}
],
"return-type": "GOutputStream",
"when": "last"
},
"get-playlist-stream": {
"args": [
{
"name": "arg0",
"type": "gchararray"
}
],
"return-type": "GOutputStream",
"when": "last"
}
}
hlssink3: Update plugin docs Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1306>
2023-09-24 11:07:46 +00:00
},
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"GstHlsSink3PlaylistType": {
"kind": "enum",
"values": [
{
"desc": "Unspecified: The tag `#EXT-X-PLAYLIST-TYPE` won't be present in the playlist during the pipeline processing.",
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 15:37:44 +00:00
"name": "unspecified",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"value": "0"
},
{
"desc": "Event: No segments will be removed from the playlist. At the end of the processing, the tag `#EXT-X-ENDLIST` is added to the playlist. The tag `#EXT-X-PLAYLIST-TYPE:EVENT` will be present in the playlist.",
"name": "event",
"value": "1"
},
{
"desc": "Vod: The playlist behaves like the `event` option (a live event), but at the end of the processing, the playlist will be set to `#EXT-X-PLAYLIST-TYPE:VOD`.",
"name": "vod",
"value": "2"
}
]
}
},
"package": "gst-plugin-hlssink3",
"source": "gst-plugin-hlssink3",
"tracers": {},
"url": "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
},
"hsv": {
Update docs
2022-10-23 17:35:14 +00:00
"description": "GStreamer plugin with HSV manipulation elements",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"elements": {
"hsvdetector": {
"author": "Julien Bardagi <julien.bardagi@gmail.com>",
"description": "Works within the HSV colorspace to mark positive pixels",
"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstHsvDetector",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"GstVideoFilter",
"GstBaseTransform",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Filter/Effect/Converter/Video",
"long-name": "HSV detector",
"pad-templates": {
"sink": {
"caps": "video/x-raw:\n format: { RGBx, xRGB, BGRx, xBGR, RGB, BGR }\n width: [ 1, 2147483647 ]\n height: [ 1, 2147483647 ]\n framerate: [ 0/1, 2147483647/1 ]\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "video/x-raw:\n format: { RGBA, ARGB, BGRA, ABGR }\n width: [ 1, 2147483647 ]\n height: [ 1, 2147483647 ]\n framerate: [ 0/1, 2147483647/1 ]\n",
"direction": "src",
"presence": "always"
}
},
"properties": {
"hue-ref": {
"blurb": "Hue reference in degrees",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "3.40282e+38",
"min": "-3.40282e+38",
"mutable": "playing",
"readable": true,
"type": "gfloat",
"writable": true
},
"hue-var": {
"blurb": "Allowed hue variation from the reference hue angle, in degrees",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "10",
"max": "180",
"min": "0",
"mutable": "playing",
"readable": true,
"type": "gfloat",
"writable": true
},
"saturation-ref": {
"blurb": "Reference saturation value",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "1",
"min": "0",
"mutable": "playing",
"readable": true,
"type": "gfloat",
"writable": true
},
"saturation-var": {
"blurb": "Allowed saturation variation from the reference value",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0.15",
"max": "1",
"min": "0",
"mutable": "playing",
"readable": true,
"type": "gfloat",
"writable": true
},
"value-ref": {
"blurb": "Reference value value",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "1",
"min": "0",
"mutable": "playing",
"readable": true,
"type": "gfloat",
"writable": true
},
"value-var": {
"blurb": "Allowed value variation from the reference value",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0.3",
"max": "1",
"min": "0",
"mutable": "playing",
"readable": true,
"type": "gfloat",
"writable": true
}
},
"rank": "none"
},
"hsvfilter": {
"author": "Julien Bardagi <julien.bardagi@gmail.com>",
fix typos Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1170>
2023-04-10 11:34:46 +00:00
"description": "Works within the HSV colorspace to apply transformations to incoming frames",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstHsvFilter",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"GstVideoFilter",
"GstBaseTransform",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Filter/Effect/Converter/Video",
"long-name": "HSV filter",
"pad-templates": {
"sink": {
"caps": "video/x-raw:\n format: { RGBx, xRGB, BGRx, xBGR, RGBA, ARGB, BGRA, ABGR, RGB, BGR }\n width: [ 1, 2147483647 ]\n height: [ 1, 2147483647 ]\n framerate: [ 0/1, 2147483647/1 ]\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "video/x-raw:\n format: { RGBx, xRGB, BGRx, xBGR, RGBA, ARGB, BGRA, ABGR, RGB, BGR }\n width: [ 1, 2147483647 ]\n height: [ 1, 2147483647 ]\n framerate: [ 0/1, 2147483647/1 ]\n",
"direction": "src",
"presence": "always"
}
},
"properties": {
"hue-shift": {
"blurb": "Hue shifting in degrees",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "3.40282e+38",
"min": "-3.40282e+38",
"mutable": "playing",
"readable": true,
"type": "gfloat",
"writable": true
},
"saturation-mul": {
"blurb": "Saturation multiplier to apply to the saturation value (before offset)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "1",
"max": "3.40282e+38",
"min": "-3.40282e+38",
"mutable": "playing",
"readable": true,
"type": "gfloat",
"writable": true
},
"saturation-off": {
"blurb": "Saturation offset to add to the saturation value (after multiplier)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "3.40282e+38",
"min": "-3.40282e+38",
"mutable": "playing",
"readable": true,
"type": "gfloat",
"writable": true
},
"value-mul": {
"blurb": "Value multiplier to apply to the value (before offset)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "1",
"max": "3.40282e+38",
"min": "-3.40282e+38",
"mutable": "playing",
"readable": true,
"type": "gfloat",
"writable": true
},
"value-off": {
"blurb": "Value offset to add to the value (after multiplier)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "3.40282e+38",
"min": "-3.40282e+38",
"mutable": "playing",
"readable": true,
"type": "gfloat",
"writable": true
}
},
"rank": "none"
}
},
"filename": "gsthsv",
"license": "MIT/X11",
"other-types": {},
"package": "gst-plugin-hsv",
"source": "gst-plugin-hsv",
"tracers": {},
"url": "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
},
Update docs
2022-10-23 17:35:14 +00:00
"json": {
"description": "GStreamer JSON Plugin",
"elements": {
"jsongstenc": {
"author": "Mathieu Duponchelle <mathieu@centricular.com>",
"description": "Wraps buffers containing any valid top-level JSON structures into higher level JSON objects, and outputs those as ndjson",
"hierarchy": [
"GstJsonGstEnc",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Encoder/JSON",
"long-name": "GStreamer buffers to JSON",
"pad-templates": {
"sink": {
"caps": "application/x-json:\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "application/x-json:\n",
"direction": "src",
"presence": "always"
}
},
"rank": "none"
},
"jsongstparse": {
"author": "Mathieu Duponchelle <mathieu@centricular.com>",
"description": "Parses ndjson as output by jsongstenc",
"hierarchy": [
"GstJsonGstParse",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Parser/JSON",
"long-name": "JSON GStreamer parser",
"pad-templates": {
"sink": {
"caps": "ANY",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "application/x-json:\n",
"direction": "src",
"presence": "always"
}
},
"rank": "primary"
}
},
"filename": "gstjson",
"license": "MPL",
"other-types": {},
"package": "gst-plugin-json",
"source": "gst-plugin-json",
"tracers": {},
"url": "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
},
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"lewton": {
Update docs
2022-10-23 17:35:14 +00:00
"description": "GStreamer lewton Vorbis Decoder Plugin",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"elements": {
"lewtondec": {
"author": "Sebastian Dröge <sebastian@centricular.com>",
"description": "lewton Vorbis decoder",
"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstLewtonDec",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"GstAudioDecoder",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Decoder/Audio",
"long-name": "lewton Vorbis decoder",
"pad-templates": {
"sink": {
"caps": "audio/x-vorbis:\n",
"direction": "sink",
"presence": "always"
},
"src": {
plugins, examples, tutorials: Use AudioCapsBuilder and VideoCapsBuilder Simplify caps creation code
2022-10-13 18:02:04 +00:00
"caps": "audio/x-raw:\n rate: [ 1, 2147483647 ]\n channels: [ 1, 255 ]\n layout: interleaved\n format: F32LE\n",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"direction": "src",
"presence": "always"
}
},
"rank": "marginal"
}
},
"filename": "gstlewton",
"license": "MIT/X11",
"other-types": {},
"package": "gst-plugin-lewton",
"source": "gst-plugin-lewton",
"tracers": {},
"url": "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
ndi: Add NDI plugin to the docs
2022-10-12 18:39:55 +00:00
},
Add livesync plugin It attempts to produce a (nearly) gapless live stream by synchronizing its output to the running time and forwarding the next input buffer if its start is (nearly) flush with the end of the last output buffer. If the input buffer is missing or too far in the future, it duplicates the last output buffer with adjusted timestamps. If it is operating on a raw audio stream, it will fill duplicate buffers with silence. If an input buffer arrives too late, it is thrown away. If the last input buffer was accepted too long ago (according to `late-threshold`), a late input buffer is accepted anyway, but immediately considered a duplicate. Due to the silence-filling, this has no effect on audio, but video gets a "slideshow" effect instead of freezing completely. The "many-repeats" property will be notified when this element has recently duplicated a lot of buffers or recovered from such a state. Co-authored-by: Vivia Nikolaidou <vivia@ahiru.eu> Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/708>
2022-10-12 13:35:11 +00:00
"livesync": {
"description": "Livesync Plugin",
"elements": {
"livesync": {
"author": "Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>",
"description": "Outputs livestream, inserting gap frames when input lags",
"hierarchy": [
"GstLiveSync",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Filter",
"long-name": "Live Synchronizer",
"pad-templates": {
"sink": {
"caps": "ANY",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "ANY",
"direction": "src",
"presence": "always"
}
},
"properties": {
"drop": {
"blurb": "Number of incoming frames dropped",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "18446744073709551615",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint64",
"writable": false
},
"duplicate": {
"blurb": "Number of outgoing frames duplicated",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "18446744073709551615",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint64",
"writable": false
},
"in": {
"blurb": "Number of incoming frames accepted",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "18446744073709551615",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint64",
"writable": false
},
"late-threshold": {
"blurb": "Maximum time spent (in nanoseconds) before accepting one late buffer; -1 = never",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "2000000000",
"max": "18446744073709551615",
livesync: fix late-threshold property min value The code is handling 0 as "always over threshold" but it was not possible to set the property to 0. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1049>
2023-01-16 15:42:56 +00:00
"min": "0",
Add livesync plugin It attempts to produce a (nearly) gapless live stream by synchronizing its output to the running time and forwarding the next input buffer if its start is (nearly) flush with the end of the last output buffer. If the input buffer is missing or too far in the future, it duplicates the last output buffer with adjusted timestamps. If it is operating on a raw audio stream, it will fill duplicate buffers with silence. If an input buffer arrives too late, it is thrown away. If the last input buffer was accepted too long ago (according to `late-threshold`), a late input buffer is accepted anyway, but immediately considered a duplicate. Due to the silence-filling, this has no effect on audio, but video gets a "slideshow" effect instead of freezing completely. The "many-repeats" property will be notified when this element has recently duplicated a lot of buffers or recovered from such a state. Co-authored-by: Vivia Nikolaidou <vivia@ahiru.eu> Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/708>
2022-10-12 13:35:11 +00:00
"mutable": "playing",
"readable": true,
"type": "guint64",
"writable": true
},
"latency": {
"blurb": "Additional latency to allow upstream to take longer to produce buffers for the current position (in nanoseconds)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "9223372036854775807",
"min": "0",
"mutable": "playing",
"readable": true,
"type": "guint64",
"writable": true
},
"out": {
"blurb": "Number of outgoing frames produced",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "18446744073709551615",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint64",
"writable": false
},
"single-segment": {
"blurb": "Timestamp buffers and eat segments so as to appear as one segment",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "ready",
"readable": true,
"type": "gboolean",
"writable": true
livesync: Add sync property for allowing to output buffers as soon as they arrive By default livesync will wait for each buffer on the clock. If sync is set to false, it will output buffers immediately once they're available and only waits on the clock for outputting gap filler buffers. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1635>
2024-06-26 08:53:40 +00:00
},
"sync": {
"blurb": "Synchronize buffers to the clock",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "true",
"mutable": "ready",
"readable": true,
"type": "gboolean",
"writable": true
Add livesync plugin It attempts to produce a (nearly) gapless live stream by synchronizing its output to the running time and forwarding the next input buffer if its start is (nearly) flush with the end of the last output buffer. If the input buffer is missing or too far in the future, it duplicates the last output buffer with adjusted timestamps. If it is operating on a raw audio stream, it will fill duplicate buffers with silence. If an input buffer arrives too late, it is thrown away. If the last input buffer was accepted too long ago (according to `late-threshold`), a late input buffer is accepted anyway, but immediately considered a duplicate. Due to the silence-filling, this has no effect on audio, but video gets a "slideshow" effect instead of freezing completely. The "many-repeats" property will be notified when this element has recently duplicated a lot of buffers or recovered from such a state. Co-authored-by: Vivia Nikolaidou <vivia@ahiru.eu> Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/708>
2022-10-12 13:35:11 +00:00
}
},
"rank": "none"
}
},
"filename": "gstlivesync",
"license": "MPL",
"other-types": {},
"package": "gst-plugin-livesync",
"source": "gst-plugin-livesync",
"tracers": {},
"url": "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
},
gst-plugin-mp4: Add new MP4 plugin with a non-fragmented MP4 muxer
2022-11-07 18:41:50 +00:00
"mp4": {
"description": "GStreamer Rust MP4 Plugin",
"elements": {
"isomp4mux": {
"author": "Sebastian Dröge <sebastian@centricular.com>",
"description": "ISO MP4 muxer",
"hierarchy": [
"GstISOMP4Mux",
"GstRsMP4Mux",
"GstAggregator",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Codec/Muxer",
"pad-templates": {
"sink_%%u": {
mp4/fmp4: support flac inside the iso (f)mp4 container Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1401>
2023-11-28 17:53:10 +00:00
"caps": "video/x-h264:\n stream-format: { (string)avc, (string)avc3 }\n alignment: au\n width: [ 1, 65535 ]\n height: [ 1, 65535 ]\nvideo/x-h265:\n stream-format: { (string)hvc1, (string)hev1 }\n alignment: au\n width: [ 1, 65535 ]\n height: [ 1, 65535 ]\nvideo/x-vp8:\n width: [ 1, 65535 ]\n height: [ 1, 65535 ]\nvideo/x-vp9:\n profile: { (string)0, (string)1, (string)2, (string)3 }\n chroma-format: { (string)4:2:0, (string)4:2:2, (string)4:4:4 }\n bit-depth-luma: { (uint)8, (uint)10, (uint)12 }\nbit-depth-chroma: { (uint)8, (uint)10, (uint)12 }\n width: [ 1, 65535 ]\n height: [ 1, 65535 ]\nvideo/x-av1:\n stream-format: obu-stream\n alignment: tu\n profile: { (string)main, (string)high, (string)professional }\n chroma-format: { (string)4:0:0, (string)4:2:0, (string)4:2:2, (string)4:4:4 }\n bit-depth-luma: { (uint)8, (uint)10, (uint)12 }\nbit-depth-chroma: { (uint)8, (uint)10, (uint)12 }\n width: [ 1, 65535 ]\n height: [ 1, 65535 ]\naudio/mpeg:\n mpegversion: 4\n stream-format: raw\n channels: [ 1, 65535 ]\n rate: [ 1, 2147483647 ]\naudio/x-opus:\nchannel-mapping-family: [ 0, 255 ]\n channels: [ 1, 8 ]\n rate: [ 1, 2147483647 ]\naudio/x-flac:\n framed: true\n channels: [ 1, 8 ]\n rate: [ 1, 655350 ]\n",
gst-plugin-mp4: Add new MP4 plugin with a non-fragmented MP4 muxer
2022-11-07 18:41:50 +00:00
"direction": "sink",
"presence": "request",
"type": "GstRsMP4MuxPad"
},
"src": {
"caps": "video/quicktime:\n variant: iso\n",
"direction": "src",
"presence": "always"
mp4: Add ONVIF non-fragmented MP4 muxer Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/974>
2022-11-09 07:02:35 +00:00
}
},
"rank": "marginal"
},
"onvifmp4mux": {
"author": "Sebastian Dröge <sebastian@centricular.com>",
"description": "ONVIF MP4 muxer",
"hierarchy": [
"GstONVIFMP4Mux",
"GstRsMP4Mux",
"GstAggregator",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Codec/Muxer",
"pad-templates": {
"sink_%%u": {
"caps": "video/x-h264:\n stream-format: { (string)avc, (string)avc3 }\n alignment: au\n width: [ 1, 65535 ]\n height: [ 1, 65535 ]\nvideo/x-h265:\n stream-format: { (string)hvc1, (string)hev1 }\n alignment: au\n width: [ 1, 65535 ]\n height: [ 1, 65535 ]\nimage/jpeg:\n width: [ 1, 65535 ]\n height: [ 1, 65535 ]\naudio/mpeg:\n mpegversion: 4\n stream-format: raw\n channels: [ 1, 65535 ]\n rate: [ 1, 2147483647 ]\naudio/x-alaw:\n channels: [ 1, 2 ]\n rate: [ 1, 2147483647 ]\naudio/x-mulaw:\n channels: [ 1, 2 ]\n rate: [ 1, 2147483647 ]\naudio/x-adpcm:\n layout: g726\n channels: 1\n rate: 8000\n bitrate: { (int)16000, (int)24000, (int)32000, (int)40000 }\napplication/x-onvif-metadata:\n parsed: true\n",
"direction": "sink",
"presence": "request",
"type": "GstRsMP4MuxPad"
},
"src": {
"caps": "video/quicktime:\n variant: iso\n",
"direction": "src",
"presence": "always"
gst-plugin-mp4: Add new MP4 plugin with a non-fragmented MP4 muxer
2022-11-07 18:41:50 +00:00
}
},
"rank": "marginal"
}
},
"filename": "gstmp4",
"license": "MPL",
"other-types": {
"GstRsMP4Mux": {
"hierarchy": [
"GstRsMP4Mux",
"GstAggregator",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"kind": "object",
"properties": {
"interleave-bytes": {
"blurb": "Interleave between streams in bytes",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "18446744073709551615",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint64",
"writable": true
},
"interleave-time": {
"blurb": "Interleave between streams in nanoseconds",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "500000000",
"max": "18446744073709551615",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint64",
"writable": true
},
"movie-timescale": {
"blurb": "Timescale to use for the movie (units per second, 0 is automatic)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "-1",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint",
"writable": true
}
}
},
"GstRsMP4MuxPad": {
"hierarchy": [
"GstRsMP4MuxPad",
"GstAggregatorPad",
"GstPad",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"kind": "object",
"properties": {
"trak-timescale": {
"blurb": "Timescale to use for the track (units per second, 0 is automatic)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "-1",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint",
"writable": true
}
}
}
},
"package": "gst-plugin-mp4",
"source": "gst-plugin-mp4",
"tracers": {},
"url": "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
},
net: New mpegtslive element This element allows wrapping an existing live "mpeg-ts source" (udpsrc, srtsrc,...) and providing a clock based on the actual PCR of the stream. Combined with `tsdemux ignore-pcr=True` downstream of it, this allows playing back the content at the same rate as the (remote) provider **and** not modify the original timestamps. Co-authored-by: Sebastian Dröge <slomo@coaxion.net> Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1640>
2024-06-10 13:37:30 +00:00
"mpegtslive": {
"description": "GStreamer MPEG-TS Live sources",
"elements": {
"mpegtslivesrc": {
"author": "Edward Hervey <edward@centricular.com>",
"description": "Wrap MPEG-TS sources and provide a live clock",
"hierarchy": [
"GstMpegTsLiveSource",
"GstBin",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"interfaces": [
"GstChildProxy"
],
"klass": "Network",
"pad-templates": {
"src": {
"caps": "ANY",
"direction": "src",
"presence": "always"
}
},
"properties": {
"source": {
"blurb": "Source element",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"mutable": "ready",
"readable": true,
"type": "GstElement",
"writable": true
},
"window-size": {
"blurb": "The size of the window used to calculate rate and offset",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "32",
"max": "1024",
"min": "2",
"mutable": "null",
"readable": true,
"type": "gint",
"writable": true
}
},
"rank": "none"
}
},
"filename": "gstmpegtslive",
"license": "MPL",
"other-types": {},
"package": "gst-plugin-mpegtslive",
"source": "gst-plugin-mpegtslive",
"tracers": {},
"url": "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
},
ndi: Add NDI plugin to the docs
2022-10-12 18:39:55 +00:00
"ndi": {
Update docs
2022-10-23 17:35:14 +00:00
"description": "GStreamer NewTek NDI Plugin",
ndi: Add NDI plugin to the docs
2022-10-12 18:39:55 +00:00
"elements": {
"ndisink": {
"author": "Sebastian Dröge <sebastian@centricular.com>",
"description": "NewTek NDI Sink",
"hierarchy": [
"GstNdiSink",
"GstBaseSink",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Sink/Audio/Video",
"long-name": "NewTek NDI Sink",
"pad-templates": {
"sink": {
"caps": "video/x-raw:\n format: { UYVY, I420, NV12, NV21, YV12, BGRA, BGRx, RGBA, RGBx }\n width: [ 1, 2147483647 ]\n height: [ 1, 2147483647 ]\n framerate: [ 0/1, 2147483647/1 ]\naudio/x-raw:\n format: F32LE\n rate: [ 1, 2147483647 ]\n channels: [ 1, 2147483647 ]\n layout: interleaved\n",
"direction": "sink",
"presence": "always"
}
},
"properties": {
"ndi-name": {
"blurb": "NDI Name to use",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
}
},
"rank": "none"
},
"ndisinkcombiner": {
"author": "Sebastian Dröge <sebastian@centricular.com>",
"description": "NewTek NDI sink audio/video combiner",
"hierarchy": [
"GstNdiSinkCombiner",
"GstAggregator",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Combiner/Audio/Video",
"long-name": "NewTek NDI Sink Combiner",
"pad-templates": {
"audio": {
"caps": "audio/x-raw:\n rate: [ 1, 2147483646 ]\n channels: [ 1, 2147483647 ]\n layout: interleaved\n format: F32LE\n",
"direction": "sink",
"presence": "request",
"type": "GstAggregatorPad"
},
"src": {
"caps": "video/x-raw:\n format: { UYVY, I420, NV12, NV21, YV12, BGRA, BGRx, RGBA, RGBx }\n width: [ 1, 2147483647 ]\n height: [ 1, 2147483647 ]\n framerate: [ 1/2147483647, 2147483646/1 ]\n",
"direction": "src",
"presence": "always",
"type": "GstAggregatorPad"
},
"video": {
"caps": "video/x-raw:\n format: { UYVY, I420, NV12, NV21, YV12, BGRA, BGRx, RGBA, RGBx }\n width: [ 1, 2147483647 ]\n height: [ 1, 2147483647 ]\n framerate: [ 1/2147483647, 2147483646/1 ]\n",
"direction": "sink",
"presence": "always",
"type": "GstAggregatorPad"
}
},
"rank": "none"
},
"ndisrc": {
"author": "Ruben Gonzalez <rubenrua@teltek.es>, Daniel Vilar <daniel.peiteado@teltek.es>, Sebastian Dröge <sebastian@centricular.com>",
"description": "NewTek NDI Source",
"hierarchy": [
"GstNdiSrc",
"GstBaseSrc",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Source/Audio/Video/Network",
"long-name": "NewTek NDI Source",
"pad-templates": {
"src": {
"caps": "application/x-ndi:\n",
"direction": "src",
"presence": "always"
}
},
"properties": {
"bandwidth": {
"blurb": "Bandwidth, -10 metadata-only, 10 audio-only, 100 highest",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "100",
"max": "100",
"min": "-10",
"mutable": "null",
"readable": true,
"type": "gint",
"writable": true
},
"color-format": {
"blurb": "Receive color format",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "uyvy-bgra (1)",
"mutable": "null",
"readable": true,
"type": "GstNdiRecvColorFormat",
"writable": true
},
"connect-timeout": {
"blurb": "Connection timeout in ms",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "10000",
"max": "-1",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
"max-queue-length": {
"blurb": "Maximum receive queue length",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "10",
"max": "-1",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
"ndi-name": {
"blurb": "NDI stream name of the sender",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
"receiver-ndi-name": {
"blurb": "NDI stream name of this receiver",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
"timeout": {
"blurb": "Receive timeout in ms",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "5000",
"max": "-1",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
"timestamp-mode": {
"blurb": "Timestamp information to use for outgoing PTS",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "auto (0)",
"mutable": "null",
"readable": true,
"type": "GstNdiTimestampMode",
"writable": true
},
"url-address": {
"blurb": "URL/address and port of the sender, e.g. 127.0.0.1:5961",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
}
},
"rank": "none"
},
"ndisrcdemux": {
"author": "Sebastian Dröge <sebastian@centricular.com>",
"description": "NewTek NDI source demuxer",
"hierarchy": [
"GstNdiSrcDemux",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Demuxer/Audio/Video",
"long-name": "NewTek NDI Source Demuxer",
"pad-templates": {
"audio": {
"caps": "ANY",
"direction": "src",
"presence": "sometimes"
},
"sink": {
"caps": "application/x-ndi:\n",
"direction": "sink",
"presence": "always"
},
"video": {
"caps": "ANY",
"direction": "src",
"presence": "sometimes"
}
},
"rank": "primary"
}
},
"filename": "gstndi",
"license": "MPL",
"other-types": {
"GstNdiRecvColorFormat": {
"kind": "enum",
"values": [
{
"desc": "BGRX or BGRA",
"name": "bgrx-bgra",
"value": "0"
},
{
"desc": "UYVY or BGRA",
"name": "uyvy-bgra",
"value": "1"
},
{
"desc": "RGBX or RGBA",
"name": "rgbx-rgba",
"value": "2"
},
{
"desc": "UYVY or RGBA",
"name": "uyvy-rgba",
"value": "3"
},
{
"desc": "Fastest",
"name": "fastest",
"value": "4"
},
{
"desc": "Best",
"name": "best",
"value": "5"
}
]
},
"GstNdiTimestampMode": {
"kind": "enum",
"values": [
{
"desc": "Auto",
"name": "auto",
"value": "0"
},
{
"desc": "Receive Time / Timecode",
"name": "receive-time-vs-timecode",
"value": "1"
},
{
"desc": "Receive Time / Timestamp",
"name": "receive-time-vs-timestamp",
"value": "2"
},
{
"desc": "NDI Timecode",
"name": "timecode",
"value": "3"
},
{
"desc": "NDI Timestamp",
"name": "timestamp",
"value": "4"
},
{
"desc": "Receive Time",
"name": "receive-time",
"value": "5"
}
]
}
},
"package": "gst-plugin-ndi",
"source": "gst-plugin-ndi",
"tracers": {},
"url": "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
},
originalbuffer: Pair of elements to keep and restore original buffer The goal is to be able to get back the original buffer after performing analysis on a transformed version. Then put the various GstMeta back on the original buffer. An example pipeline would be .. ! originalbuffersave ! videoscale ! analysis ! originalbufferestore ! draw_overlay ! sink Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1428>
2024-01-09 14:36:25 +00:00
"originalbuffer": {
"description": "GStreamer Origin buffer meta Plugin",
"elements": {
"originalbufferrestore": {
"author": "Olivier Crête <olivier.crete@collabora.com>",
"description": "Restores a reference to the buffer in a meta",
"hierarchy": [
"GstOriginalBufferRestore",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Generic",
"pad-templates": {
"sink": {
"caps": "ANY",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "ANY",
"direction": "src",
"presence": "always"
}
},
"rank": "none"
},
"originalbuffersave": {
"author": "Olivier Crête <olivier.crete@collabora.com>",
"description": "Saves a reference to the buffer in a meta",
"hierarchy": [
"GstOriginalBufferSave",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Generic",
"pad-templates": {
"sink": {
"caps": "ANY",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "ANY",
"direction": "src",
"presence": "always"
}
},
"rank": "none"
}
},
"filename": "gstoriginalbuffer",
"license": "MPL",
"other-types": {},
"package": "gst-plugin-originalbuffer",
"source": "gst-plugin-originalbuffer",
"tracers": {},
"url": "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
},
docs: Add documentation for gst-plugin-quinn Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1036>
2024-05-01 16:08:32 +00:00
"quinn": {
"description": "GStreamer Plugin for QUIC",
"elements": {
"quinnquicsink": {
"author": "Sanchayan Maity <sanchayan@asymptotic.io>",
"description": "Send data over the network via QUIC",
"hierarchy": [
net/quinn: Use camel case acronym Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1558>
2024-05-02 12:35:15 +00:00
"GstQuinnQuicSink",
docs: Add documentation for gst-plugin-quinn Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1036>
2024-05-01 16:08:32 +00:00
"GstBaseSink",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Source/Network/QUIC",
"pad-templates": {
"sink": {
"caps": "ANY",
"direction": "sink",
"presence": "always"
}
},
"properties": {
net/quinn: Make QUIC role configurable Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1575>
2024-05-15 08:53:10 +00:00
"address": {
"blurb": "Address of the QUIC server e.g. 127.0.0.1",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "127.0.0.1",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
docs: Add documentation for gst-plugin-quinn Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1036>
2024-05-01 16:08:32 +00:00
"alpn-protocols": {
"blurb": "QUIC connection Application-Layer Protocol Negotiation (ALPN) values",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"mutable": "null",
"readable": true,
"type": "GstValueArray",
"writable": true
},
net/quinn: Make QUIC role configurable Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1575>
2024-05-15 08:53:10 +00:00
"bind-address": {
"blurb": "Address to bind QUIC client e.g. 0.0.0.0",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0.0.0.0",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
"bind-port": {
"blurb": "Port to bind QUIC client e.g. 5001",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "65535",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
docs: Add documentation for gst-plugin-quinn Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1036>
2024-05-01 16:08:32 +00:00
"certificate-file": {
"blurb": "Path to certificate chain in single file",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
"client-address": {
"blurb": "Address to be used by this QUIC client e.g. 127.0.0.1",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "127.0.0.1",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
"client-port": {
"blurb": "Port to be used by this QUIC client e.g. 5001",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "5001",
"max": "65535",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
net/quinn: Enable client to keep QUIC conn alive Co-authored-by: Felician Nemeth <nemethf@tmit.bme.hu> Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1568>
2024-05-10 09:30:16 +00:00
},
net/quinn: Allow setting some parameters from TransportConfig As of now, we expose the below four properties from `TransportConfig`. - Initial MTU - Minimum MTU - Datagram receive buffer size - Datagram send buffer size Maximum UDP payload size from `EndpointConfig` and upper bound from `MtuDiscoveryConfig` are also exposed as properties. See the below documentation for further details. - https://docs.rs/quinn/latest/quinn/struct.TransportConfig.html - https://docs.rs/quinn/latest/quinn/struct.MtuDiscoveryConfig.html - https://docs.rs/quinn/latest/quinn/struct.EndpointConfig.html While at it, also clean up passing function parameters to the functions in utils.rs. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1613>
2024-06-13 16:06:50 +00:00
"datagram-receive-buffer-size": {
"blurb": "Maximum number of incoming application datagram bytes to buffer",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "1250000",
"max": "18446744073709551615",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint64",
"writable": true
},
"datagram-send-buffer-size": {
"blurb": "Maximum number of outgoing application datagram bytes to buffer",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "1048576",
"max": "18446744073709551615",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint64",
"writable": true
},
net/quinn: Allow dropping buffers when buffer size exceeds maximum datagram size Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1613>
2024-06-15 08:22:16 +00:00
"drop-buffer-for-datagram": {
"blurb": "Drop buffers when using datagram if buffer size > max datagram size",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "null",
"readable": true,
"type": "gboolean",
"writable": true
},
net/quinn: Allow setting some parameters from TransportConfig As of now, we expose the below four properties from `TransportConfig`. - Initial MTU - Minimum MTU - Datagram receive buffer size - Datagram send buffer size Maximum UDP payload size from `EndpointConfig` and upper bound from `MtuDiscoveryConfig` are also exposed as properties. See the below documentation for further details. - https://docs.rs/quinn/latest/quinn/struct.TransportConfig.html - https://docs.rs/quinn/latest/quinn/struct.MtuDiscoveryConfig.html - https://docs.rs/quinn/latest/quinn/struct.EndpointConfig.html While at it, also clean up passing function parameters to the functions in utils.rs. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1613>
2024-06-13 16:06:50 +00:00
"initial-mtu": {
"blurb": "Initial value to be used as the maximum UDP payload size",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "1200",
"max": "-1",
"min": "1200",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
net/quinn: Enable client to keep QUIC conn alive Co-authored-by: Felician Nemeth <nemethf@tmit.bme.hu> Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1568>
2024-05-10 09:30:16 +00:00
"keep-alive-interval": {
"blurb": "Keeps QUIC connection alive by periodically pinging the server. Value set in ms, 0 disables this feature",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "18446744073709551615",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint64",
"writable": true
docs: Add documentation for gst-plugin-quinn Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1036>
2024-05-01 16:08:32 +00:00
},
net/quinn: Allow setting some parameters from TransportConfig As of now, we expose the below four properties from `TransportConfig`. - Initial MTU - Minimum MTU - Datagram receive buffer size - Datagram send buffer size Maximum UDP payload size from `EndpointConfig` and upper bound from `MtuDiscoveryConfig` are also exposed as properties. See the below documentation for further details. - https://docs.rs/quinn/latest/quinn/struct.TransportConfig.html - https://docs.rs/quinn/latest/quinn/struct.MtuDiscoveryConfig.html - https://docs.rs/quinn/latest/quinn/struct.EndpointConfig.html While at it, also clean up passing function parameters to the functions in utils.rs. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1613>
2024-06-13 16:06:50 +00:00
"max-udp-payload-size": {
"blurb": "Maximum UDP payload size accepted from peers (excluding UDP and IP overhead)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "65527",
"max": "65527",
"min": "1200",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
"min-mtu": {
"blurb": "Maximum UDP payload size guaranteed to be supported by the network, must be <= initial-mtu",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "1200",
"max": "-1",
"min": "1200",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
net/quinn: Make QUIC role configurable Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1575>
2024-05-15 08:53:10 +00:00
"port": {
"blurb": "Port of the QUIC server e.g. 5000",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "5000",
"max": "65535",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
docs: Add documentation for gst-plugin-quinn Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1036>
2024-05-01 16:08:32 +00:00
"private-key-file": {
"blurb": "Path to a PKCS8 or RSA private key file",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
net/quinn: Make QUIC role configurable Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1575>
2024-05-15 08:53:10 +00:00
"role": {
"blurb": "QUIC connection role to use.",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "client (1)",
"mutable": "null",
"readable": true,
"type": "GstQuinnQuicRole",
"writable": true
},
docs: Add documentation for gst-plugin-quinn Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1036>
2024-05-01 16:08:32 +00:00
"secure-connection": {
"blurb": "Use certificates for QUIC connection. False: Insecure connection, True: Secure connection.",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "true",
"mutable": "null",
"readable": true,
"type": "gboolean",
"writable": true
},
"server-address": {
"blurb": "Address of the QUIC server to connect to e.g. 127.0.0.1",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "127.0.0.1",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
"server-name": {
net/quinn: Make QUIC role configurable Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1575>
2024-05-15 08:53:10 +00:00
"blurb": "Name of the QUIC server which is in server certificate in case of server role",
docs: Add documentation for gst-plugin-quinn Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1036>
2024-05-01 16:08:32 +00:00
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "localhost",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
"server-port": {
"blurb": "Port of the QUIC server to connect to e.g. 5000",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "5000",
"max": "65535",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
net/quinn: Add stats property for connection statistics Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1613>
2024-06-14 07:55:39 +00:00
"stats": {
"blurb": "Connection statistics",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "stats;",
"mutable": "null",
"readable": true,
"type": "GstStructure",
"writable": false
},
docs: Add documentation for gst-plugin-quinn Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1036>
2024-05-01 16:08:32 +00:00
"timeout": {
"blurb": "Value in seconds to timeout QUIC endpoint requests (0 = No timeout).",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "15",
"max": "3600",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
net/quinn: Allow setting some parameters from TransportConfig As of now, we expose the below four properties from `TransportConfig`. - Initial MTU - Minimum MTU - Datagram receive buffer size - Datagram send buffer size Maximum UDP payload size from `EndpointConfig` and upper bound from `MtuDiscoveryConfig` are also exposed as properties. See the below documentation for further details. - https://docs.rs/quinn/latest/quinn/struct.TransportConfig.html - https://docs.rs/quinn/latest/quinn/struct.MtuDiscoveryConfig.html - https://docs.rs/quinn/latest/quinn/struct.EndpointConfig.html While at it, also clean up passing function parameters to the functions in utils.rs. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1613>
2024-06-13 16:06:50 +00:00
"upper-bound-mtu": {
"blurb": "Upper bound to the max UDP payload size that MTU discovery will search for",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "1452",
"max": "65527",
"min": "1452",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
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"use-datagram": {
"blurb": "Use datagram for lower latency, unreliable messaging",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "null",
"readable": true,
"type": "gboolean",
"writable": true
}
},
"rank": "marginal"
},
"quinnquicsrc": {
"author": "Sanchayan Maity <sanchayan@asymptotic.io>",
"description": "Receive data over the network via QUIC",
"hierarchy": [
net/quinn: Use camel case acronym Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1558>
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"GstQuinnQuicSrc",
docs: Add documentation for gst-plugin-quinn Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1036>
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"GstBaseSrc",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Source/Network/QUIC",
"pad-templates": {
"src": {
"caps": "ANY",
"direction": "src",
"presence": "always"
}
},
"properties": {
net/quinn: Make QUIC role configurable Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1575>
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"address": {
"blurb": "Address of the QUIC server e.g. 127.0.0.1",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "127.0.0.1",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
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"alpn-protocols": {
"blurb": "QUIC connection Application-Layer Protocol Negotiation (ALPN) values",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"mutable": "null",
"readable": true,
"type": "GstValueArray",
"writable": true
},
net/quinn: Make QUIC role configurable Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1575>
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"bind-address": {
"blurb": "Address to bind QUIC client e.g. 0.0.0.0",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0.0.0.0",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
"bind-port": {
"blurb": "Port to bind QUIC client e.g. 5001",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "65535",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
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"caps": {
"blurb": "The caps of the source pad",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "ANY",
"mutable": "null",
"readable": true,
"type": "GstCaps",
"writable": true
},
"certificate-file": {
"blurb": "Path to certificate chain in single file",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
net/quinn: Allow setting some parameters from TransportConfig As of now, we expose the below four properties from `TransportConfig`. - Initial MTU - Minimum MTU - Datagram receive buffer size - Datagram send buffer size Maximum UDP payload size from `EndpointConfig` and upper bound from `MtuDiscoveryConfig` are also exposed as properties. See the below documentation for further details. - https://docs.rs/quinn/latest/quinn/struct.TransportConfig.html - https://docs.rs/quinn/latest/quinn/struct.MtuDiscoveryConfig.html - https://docs.rs/quinn/latest/quinn/struct.EndpointConfig.html While at it, also clean up passing function parameters to the functions in utils.rs. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1613>
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"datagram-receive-buffer-size": {
"blurb": "Maximum number of incoming application datagram bytes to buffer",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "1250000",
"max": "18446744073709551615",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint64",
"writable": true
},
"datagram-send-buffer-size": {
"blurb": "Maximum number of outgoing application datagram bytes to buffer",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "1048576",
"max": "18446744073709551615",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint64",
"writable": true
},
"initial-mtu": {
"blurb": "Initial value to be used as the maximum UDP payload size",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "1200",
"max": "-1",
"min": "1200",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
net/quinn: Make QUIC role configurable Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1575>
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"keep-alive-interval": {
"blurb": "Keeps QUIC connection alive by periodically pinging the server. Value set in ms, 0 disables this feature",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "18446744073709551615",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint64",
"writable": true
},
net/quinn: Allow setting some parameters from TransportConfig As of now, we expose the below four properties from `TransportConfig`. - Initial MTU - Minimum MTU - Datagram receive buffer size - Datagram send buffer size Maximum UDP payload size from `EndpointConfig` and upper bound from `MtuDiscoveryConfig` are also exposed as properties. See the below documentation for further details. - https://docs.rs/quinn/latest/quinn/struct.TransportConfig.html - https://docs.rs/quinn/latest/quinn/struct.MtuDiscoveryConfig.html - https://docs.rs/quinn/latest/quinn/struct.EndpointConfig.html While at it, also clean up passing function parameters to the functions in utils.rs. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1613>
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"max-udp-payload-size": {
"blurb": "Maximum UDP payload size accepted from peers (excluding UDP and IP overhead)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "65527",
"max": "65527",
"min": "1200",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
"min-mtu": {
"blurb": "Maximum UDP payload size guaranteed to be supported by the network, must be <= initial-mtu",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "1200",
"max": "-1",
"min": "1200",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
net/quinn: Make QUIC role configurable Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1575>
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"port": {
"blurb": "Port of the QUIC server e.g. 5000",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "5000",
"max": "65535",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
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"private-key-file": {
"blurb": "Path to a PKCS8 or RSA private key file",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
net/quinn: Make QUIC role configurable Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1575>
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"role": {
"blurb": "QUIC connection role to use.",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "server (0)",
"mutable": "null",
"readable": true,
"type": "GstQuinnQuicRole",
"writable": true
},
docs: Add documentation for gst-plugin-quinn Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1036>
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"secure-connection": {
"blurb": "Use certificates for QUIC connection. False: Insecure connection, True: Secure connection.",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "true",
"mutable": "null",
"readable": true,
"type": "gboolean",
"writable": true
},
"server-address": {
"blurb": "Address of the QUIC server e.g. 127.0.0.1",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "127.0.0.1",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
"server-name": {
net/quinn: Make QUIC role configurable Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1575>
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"blurb": "Name of the QUIC server which is in server certificate in case of server role",
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"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "localhost",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
"server-port": {
"blurb": "Port of the QUIC server e.g. 5000",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "5000",
"max": "65535",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
net/quinn: Add stats property for connection statistics Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1613>
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"stats": {
"blurb": "Connection statistics",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "stats;",
"mutable": "null",
"readable": true,
"type": "GstStructure",
"writable": false
},
docs: Add documentation for gst-plugin-quinn Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1036>
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"timeout": {
"blurb": "Value in seconds to timeout QUIC endpoint requests (0 = No timeout).",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "15",
"max": "3600",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
net/quinn: Allow setting some parameters from TransportConfig As of now, we expose the below four properties from `TransportConfig`. - Initial MTU - Minimum MTU - Datagram receive buffer size - Datagram send buffer size Maximum UDP payload size from `EndpointConfig` and upper bound from `MtuDiscoveryConfig` are also exposed as properties. See the below documentation for further details. - https://docs.rs/quinn/latest/quinn/struct.TransportConfig.html - https://docs.rs/quinn/latest/quinn/struct.MtuDiscoveryConfig.html - https://docs.rs/quinn/latest/quinn/struct.EndpointConfig.html While at it, also clean up passing function parameters to the functions in utils.rs. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1613>
2024-06-13 16:06:50 +00:00
},
"upper-bound-mtu": {
"blurb": "Upper bound to the max UDP payload size that MTU discovery will search for",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "1452",
"max": "65527",
"min": "1452",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
docs: Add documentation for gst-plugin-quinn Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1036>
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},
"use-datagram": {
"blurb": "Use datagram for lower latency, unreliable messaging",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "null",
"readable": true,
"type": "gboolean",
"writable": true
}
},
"rank": "marginal"
}
},
"filename": "gstquinn",
"license": "MPL",
net/quinn: Make QUIC role configurable Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1575>
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"other-types": {
"GstQuinnQuicRole": {
"kind": "enum",
"values": [
{
"desc": "Server: Act as QUIC server.",
"name": "server",
"value": "0"
},
{
"desc": "Client: Act as QUIC client.",
"name": "client",
"value": "1"
}
]
}
},
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"package": "gst-plugin-quinn",
"source": "gst-plugin-quinn",
"tracers": {},
"url": "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
},
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"raptorq": {
Update docs
2022-10-23 17:35:14 +00:00
"description": "GStreamer RaptorQ FEC Plugin",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
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"elements": {
"raptorqdec": {
"author": "Tomasz Andrzejak <andreiltd@gmail.com>",
"description": "Performs FEC using RaptorQ (RFC6681, RFC6682)",
"hierarchy": [
"GstRaptorqDec",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "RTP RaptorQ FEC Decoding",
"long-name": "RTP RaptorQ FEC Decoder",
"pad-templates": {
"fec_%%u": {
"caps": "application/x-rtp:\nraptor-scheme-id: 6\n",
"direction": "sink",
"presence": "request"
},
"sink": {
"caps": "application/x-rtp:\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "application/x-rtp:\n",
"direction": "src",
"presence": "always"
}
},
"properties": {
"media-packets-reset-threshold": {
"blurb": "This is the maximum allowed number of buffered packets, before we reset the decoder. It can only be triggered if we don't receive repair packets for too long, or packets have no valid timestamps, (0 - disable)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "5000",
"max": "-2",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint",
"writable": true
},
"repair-window-tolerance": {
"blurb": "The amount of time to add to repair-window reported by RaptorQ encoder (in ms)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "500",
"max": "-2",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint",
"writable": true
},
"stats": {
"blurb": "Various statistics",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "application/x-rtp-raptorqdec-stats, received-packets=(guint64)0, lost-packets=(guint64)0, recovered-packets=(guint64)0, buffered-media-packets=(guint64)0, buffered-repair-packets=(guint64)0;",
"mutable": "null",
"readable": true,
"type": "GstStructure",
"writable": false
}
},
"rank": "marginal"
},
"raptorqenc": {
"author": "Tomasz Andrzejak <andreiltd@gmail.com>",
"description": "Performs FEC using RaptorQ (RFC6681, RFC6682)",
"hierarchy": [
"GstRaptorqEnc",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "RTP RaptorQ FEC Encoding",
"long-name": "RTP RaptorQ FEC Encoder",
"pad-templates": {
"fec_0": {
"caps": "application/x-rtp:\n clock-rate: [ 0, 2147483647 ]\n",
"direction": "src",
"presence": "always"
},
"sink": {
"caps": "application/x-rtp:\n clock-rate: [ 0, 2147483647 ]\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "application/x-rtp:\n clock-rate: [ 0, 2147483647 ]\n",
"direction": "src",
"presence": "always"
}
},
"properties": {
"mtu": {
"blurb": "Maximum expected packet size",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "1400",
"max": "2147483647",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint",
"writable": true
},
"protected-packets": {
"blurb": "Number of packets to protect together",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "25",
"max": "-2",
"min": "1",
"mutable": "ready",
"readable": true,
"type": "guint",
"writable": true
},
"pt": {
"blurb": "The payload type of FEC packets",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "97",
"max": "255",
"min": "96",
"mutable": "ready",
"readable": true,
"type": "guint",
"writable": true
},
"repair-packets": {
"blurb": "Number of repair packets per block to send",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "5",
"max": "-2",
"min": "1",
"mutable": "ready",
"readable": true,
"type": "guint",
"writable": true
},
"repair-window": {
"blurb": "A time span in milliseconds in which repair packets are send",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "50",
"max": "-2",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint",
"writable": true
},
"symbol-size": {
"blurb": "Size of RaptorQ data unit",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "1408",
"max": "-2",
"min": "1",
"mutable": "ready",
"readable": true,
"type": "guint",
"writable": true
}
},
"rank": "marginal"
}
},
"filename": "gstraptorq",
"license": "MPL",
"other-types": {},
"package": "gst-plugin-raptorq",
"source": "gst-plugin-raptorq",
"tracers": {},
"url": "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
},
"rav1e": {
Update docs
2022-10-23 17:35:14 +00:00
"description": "GStreamer rav1e AV1 Encoder Plugin",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"elements": {
"rav1enc": {
"author": "Sebastian Dröge <sebastian@centricular.com>",
"description": "rav1e AV1 encoder",
"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstRav1Enc",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"GstVideoEncoder",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"interfaces": [
"GstPreset"
],
"klass": "Encoder/Video",
"long-name": "rav1e AV1 encoder",
"pad-templates": {
"sink": {
"caps": "video/x-raw:\n format: { I420, Y42B, Y444, I420_10LE, I422_10LE, Y444_10LE, I420_12LE, I422_12LE, Y444_12LE, GRAY8 }\n width: [ 1, 2147483647 ]\n height: [ 1, 2147483647 ]\n framerate: [ 0/1, 2147483647/1 ]\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "video/x-av1:\n stream-format: obu-stream\n alignment: tu\n",
"direction": "src",
"presence": "always"
}
},
"properties": {
"bitrate": {
rav1enc: document bitrate property unit See: https://github.com/xiph/rav1e/blob/e34e772e47b01169b6f75a4589c056624ea886a4/src/rate.rs#L365 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1667>
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"blurb": "Bitrate in bits per second",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
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"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "2147483647",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "gint",
"writable": true
},
"error-resilient": {
"blurb": "Error Resilient",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "ready",
"readable": true,
"type": "gboolean",
"writable": true
},
"low-latency": {
"blurb": "Low Latency",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "ready",
"readable": true,
"type": "gboolean",
"writable": true
},
"max-key-frame-interval": {
"blurb": "Max Key Frame Interval",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "240",
"max": "18446744073709551615",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint64",
"writable": true
},
"min-key-frame-interval": {
"blurb": "Min Key Frame Interval",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "12",
"max": "18446744073709551615",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint64",
"writable": true
},
"min-quantizer": {
"blurb": "Min Quantizer",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "255",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint",
"writable": true
},
"quantizer": {
"blurb": "Quantizer",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "100",
"max": "-1",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint",
"writable": true
},
"rdo-lookahead-frames": {
"blurb": "RDO Lookahead Frames",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "-1",
"max": "2147483647",
"min": "-1",
"mutable": "ready",
"readable": true,
"type": "gint",
"writable": true
},
"reservoir-frame-delay": {
"blurb": "Reservoir Frame Delay",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "-2147483648",
"max": "2147483647",
"min": "-2147483648",
"mutable": "ready",
"readable": true,
"type": "gint",
"writable": true
},
"speed-preset": {
"blurb": "Speed preset (10 fastest, 0 slowest)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "6",
"max": "10",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint",
"writable": true
},
"switch-frame-interval": {
"blurb": "Switch Frame Interval",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "18446744073709551615",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint64",
"writable": true
},
"threads": {
"blurb": "Threads",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "-1",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint",
"writable": true
},
"tile-cols": {
"blurb": "Tile Cols",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "-1",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint",
"writable": true
},
"tile-rows": {
"blurb": "Tile Rows",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "-1",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint",
"writable": true
},
"tiles": {
"blurb": "Tiles",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "-1",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint",
"writable": true
},
"tune": {
"blurb": "Tune",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "psychovisual (1)",
"mutable": "ready",
"readable": true,
"type": "GstRav1eEncTune",
"writable": true
}
},
"rank": "primary"
}
},
"filename": "gstrav1e",
"license": "MIT/X11",
"other-types": {
"GstRav1eEncTune": {
"kind": "enum",
"values": [
{
"desc": "Psnr",
"name": "psnr",
"value": "0"
},
{
"desc": "Psychovisual",
"name": "psychovisual",
"value": "1"
}
]
}
},
"package": "gst-plugin-rav1e",
"source": "gst-plugin-rav1e",
"tracers": {},
"url": "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
},
Update docs
2022-10-23 17:35:14 +00:00
"regex": {
"description": "GStreamer Regular Expression Plugin",
"elements": {
"regex": {
"author": "Mathieu Duponchelle <mathieu@centricular.com>",
"description": "Applies operations according to regular expressions",
"hierarchy": [
"GstRegEx",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Text/Filter",
"long-name": "Regular Expression processor",
"pad-templates": {
"sink": {
"caps": "text/x-raw:\n format: utf8\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "text/x-raw:\n format: utf8\n",
"direction": "src",
"presence": "always"
}
},
"properties": {
"commands": {
"blurb": "A set of commands to apply on input text",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"mutable": "playing",
"readable": true,
"type": "GstValueArray",
"writable": true
}
},
"rank": "none"
}
},
"filename": "gstregex",
"license": "MPL",
"other-types": {},
"package": "gst-plugin-regex",
"source": "gst-plugin-regex",
"tracers": {},
"url": "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
},
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"reqwest": {
Update docs
2022-10-23 17:35:14 +00:00
"description": "GStreamer reqwest HTTP Source Plugin",
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"elements": {
"reqwesthttpsrc": {
"author": "Sebastian Dröge <sebastian@centricular.com>",
"description": "Read stream from an HTTP/HTTPS location",
"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstReqwestHttpSrc",
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"GstPushSrc",
"GstBaseSrc",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"interfaces": [
"GstURIHandler"
],
"klass": "Source/Network/HTTP",
"long-name": "HTTP Source",
"pad-templates": {
"src": {
"caps": "ANY",
"direction": "src",
"presence": "always"
}
},
"properties": {
"compress": {
"blurb": "Allow compressed content encodings",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "ready",
"readable": true,
"type": "gboolean",
"writable": true
},
"cookies": {
"blurb": "NULL",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"mutable": "ready",
"readable": true,
"type": "GStrv",
"writable": true
},
"extra-headers": {
"blurb": "Extra headers to append to the HTTP request",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"mutable": "ready",
"readable": true,
"type": "GstStructure",
"writable": true
},
"iradio-mode": {
"blurb": "Enable internet radio mode (ask server to send shoutcast/icecast metadata interleaved with the actual stream data",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "true",
"mutable": "ready",
"readable": true,
"type": "gboolean",
"writable": true
},
"is-live": {
"blurb": "Act like a live source",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "ready",
"readable": true,
"type": "gboolean",
"writable": true
},
"keep-alive": {
"blurb": "Use HTTP persistent connections",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "true",
"mutable": "ready",
"readable": true,
"type": "gboolean",
"writable": true
},
"location": {
"blurb": "URL to read from",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
},
"proxy": {
"blurb": "HTTP proxy server URI",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
},
"proxy-id": {
"blurb": "HTTP proxy URI user id for authentication",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
},
"proxy-pw": {
"blurb": "HTTP proxy URI user password for authentication",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
},
"timeout": {
"blurb": "Value in seconds to timeout a blocking I/O (0 = No timeout).",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "15",
"max": "3600",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint",
"writable": true
},
"user-agent": {
"blurb": "Value of the User-Agent HTTP request header field",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
doc: Mark request::user-agent as doc show default
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"default": "GStreamer reqwesthttpsrc",
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"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
},
"user-id": {
"blurb": "HTTP location URI user id for authentication",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
},
"user-pw": {
"blurb": "HTTP location URI user password for authentication",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
}
},
"rank": "marginal"
}
},
"filename": "gstreqwest",
"license": "MIT/X11",
"other-types": {},
"package": "gst-plugin-reqwest",
"source": "gst-plugin-reqwest",
"tracers": {},
"url": "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
},
"rsaudiofx": {
Update docs
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"description": "GStreamer Rust Audio Effects Plugin",
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"elements": {
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"audioloudnorm": {
"author": "Sebastian Dröge <sebastian@centricular.com>",
"description": "Normalizes perceived loudness of an audio stream",
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"hierarchy": [
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"GstAudioLoudNorm",
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"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Filter/Effect/Audio",
Update docs
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"long-name": "Audio loudness normalizer",
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"pad-templates": {
"sink": {
Update docs
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"caps": "audio/x-raw:\n rate: 192000\n channels: [ 1, 2147483647 ]\n layout: interleaved\n format: F64LE\n",
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"direction": "sink",
"presence": "always"
},
"src": {
Update docs
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"caps": "audio/x-raw:\n rate: 192000\n channels: [ 1, 2147483647 ]\n layout: interleaved\n format: F64LE\n",
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"direction": "src",
"presence": "always"
}
},
Update docs
2022-10-23 17:35:14 +00:00
"properties": {
"loudness-range-target": {
"blurb": "Loudness range target in LU",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "7",
"max": "20",
"min": "1",
"mutable": "ready",
"readable": true,
"type": "gdouble",
"writable": true
},
"loudness-target": {
"blurb": "Loudness target in LUFS",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "-24",
"max": "-5",
"min": "-70",
"mutable": "ready",
"readable": true,
"type": "gdouble",
"writable": true
},
"max-true-peak": {
"blurb": "Maximum True Peak in dbTP",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "-2",
"max": "0",
"min": "-9",
"mutable": "ready",
"readable": true,
"type": "gdouble",
"writable": true
},
"offset": {
"blurb": "Offset Gain in LU",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "99",
"min": "-99",
"mutable": "ready",
"readable": true,
"type": "gdouble",
"writable": true
}
},
"rank": "none"
},
"audiornnoise": {
"author": "Philippe Normand <philn@igalia.com>",
"description": "Removes noise from an audio stream",
"hierarchy": [
"GstAudioRNNoise",
audiofx: Derive from AudioFilter where possible Saves a little bit of code. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1013>
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"GstAudioFilter",
Update docs
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"GstBaseTransform",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Filter/Effect/Audio",
"long-name": "Audio denoise",
"pad-templates": {
"sink": {
"caps": "audio/x-raw:\n rate: 48000\n channels: [ 1, 2147483647 ]\n layout: interleaved\n format: F32LE\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "audio/x-raw:\n rate: 48000\n channels: [ 1, 2147483647 ]\n layout: interleaved\n format: F32LE\n",
"direction": "src",
"presence": "always"
}
},
audiornnoise: add voice detection threshold Add a property "voice-activity-threshold". Frames where the voice detection score from the RNN is below the threshold will be completely muted. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1004>
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"properties": {
"voice-activity-threshold": {
"blurb": "Threshold of the voice activity detector below which to mute the output",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "1",
"min": "0",
"mutable": "playing",
"readable": true,
"type": "gfloat",
"writable": true
}
},
Update docs
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"rank": "none"
},
"ebur128level": {
"author": "Sebastian Dröge <sebastian@centricular.com>",
"description": "Measures different loudness metrics according to EBU R128",
"hierarchy": [
"GstEbuR128Level",
audiofx: Derive from AudioFilter where possible Saves a little bit of code. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1013>
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"GstAudioFilter",
Update docs
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"GstBaseTransform",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Filter/Analyzer/Audio",
"long-name": "EBU R128 Loudness Level Measurement",
"pad-templates": {
"sink": {
"caps": "audio/x-raw:\n rate: [ 1, 2822399 ]\n channels: [ 1, 63 ]\n layout: { (string)interleaved, (string)non-interleaved }\n format: { S16LE, S32LE, F32LE, F64LE }\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "audio/x-raw:\n rate: [ 1, 2822399 ]\n channels: [ 1, 63 ]\n layout: { (string)interleaved, (string)non-interleaved }\n format: { S16LE, S32LE, F32LE, F64LE }\n",
"direction": "src",
"presence": "always"
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}
},
"properties": {
"interval": {
"blurb": "Interval in nanoseconds for posting messages",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "1000000000",
"max": "18446744073709551614",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint64",
"writable": true
},
"mode": {
"blurb": "Selection of metrics to calculate",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "true-peak+sample-peak+loudness-range+global+short-term+momentary",
"mutable": "ready",
"readable": true,
Update docs
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"type": "GstEbuR128LevelMode",
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"writable": true
},
"post-messages": {
"blurb": "Whether to post messages on the bus for each interval",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "true",
"mutable": "playing",
"readable": true,
"type": "gboolean",
"writable": true
}
},
"rank": "none",
"signals": {
"reset": {
"action": true,
"args": [],
"return-type": "void",
"when": "last"
}
}
},
"hrtfrender": {
"author": "Tomasz Andrzejak <andreiltd@gmail.com>",
"description": "Renders spatial sounds to a given position",
"hierarchy": [
Update docs
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"GstHrtfRender",
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2022-08-25 22:30:08 +00:00
"GstBaseTransform",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Filter/Effect/Audio",
"long-name": "Head-Related Transfer Function (HRTF) renderer",
"pad-templates": {
"sink": {
plugins, examples, tutorials: Use AudioCapsBuilder and VideoCapsBuilder Simplify caps creation code
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"caps": "audio/x-raw:\n rate: [ 1, 2147483647 ]\n channels: [ 1, 64 ]\n layout: interleaved\n format: F32LE\n",
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"direction": "sink",
"presence": "always"
},
"src": {
plugins, examples, tutorials: Use AudioCapsBuilder and VideoCapsBuilder Simplify caps creation code
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"caps": "audio/x-raw:\n rate: [ 1, 2147483647 ]\n channels: 2\n layout: interleaved\n format: F32LE\n",
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"direction": "src",
"presence": "always"
}
},
"properties": {
"block-length": {
"blurb": "Block Length is the length of each slice",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "512",
"max": "18446744073709551614",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint64",
"writable": true
},
"hrir-file": {
"blurb": "Head Transform Impulse Response file location to read from",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
},
"hrir-raw": {
"blurb": "Head Transform Impulse Response raw bytes",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"mutable": "ready",
"readable": true,
"type": "GBytes",
"writable": true
},
"interpolation-steps": {
"blurb": "Interpolation Steps is the amount of slices to cut source to",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "8",
"max": "18446744073709551614",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint64",
"writable": true
},
"spatial-objects": {
"blurb": "Spatial object Metadata to apply on input channels",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"mutable": "playing",
"readable": true,
"type": "GstValueArray",
"writable": true
}
},
"rank": "none"
},
"rsaudioecho": {
"author": "Sebastian Dröge <sebastian@centricular.com>",
"description": "Adds an echo or reverb effect to an audio stream",
"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstRsAudioEcho",
audiofx: Derive from AudioFilter where possible Saves a little bit of code. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1013>
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"GstAudioFilter",
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2022-08-25 22:30:08 +00:00
"GstBaseTransform",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Filter/Effect/Audio",
"long-name": "Audio echo",
"pad-templates": {
"sink": {
"caps": "audio/x-raw:\n rate: [ 1, 2147483647 ]\n channels: [ 1, 2147483647 ]\n layout: interleaved\n format: { F32LE, F64LE }\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "audio/x-raw:\n rate: [ 1, 2147483647 ]\n channels: [ 1, 2147483647 ]\n layout: interleaved\n format: { F32LE, F64LE }\n",
"direction": "src",
"presence": "always"
}
},
"properties": {
"delay": {
"blurb": "Delay of the echo in nanoseconds",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "500000000000",
"max": "18446744073709551614",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint64",
"writable": true
},
"feedback": {
"blurb": "Amount of feedback",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "1",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "gdouble",
"writable": true
},
"intensity": {
"blurb": "Intensity of the echo",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0.5",
"max": "1",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "gdouble",
"writable": true
},
"max-delay": {
"blurb": "Maximum delay of the echo in nanoseconds (can't be changed in PLAYING or PAUSED state)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "1000000000",
"max": "18446744073709551614",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint64",
"writable": true
}
},
"rank": "none"
}
},
"filename": "gstrsaudiofx",
"license": "MPL",
"other-types": {
Update docs
2022-10-23 17:35:14 +00:00
"GstEbuR128LevelMode": {
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"kind": "flags",
"values": [
{
"desc": "Calculate momentary loudness (400ms)",
"name": "momentary",
"value": "0x00000001"
},
{
"desc": "Calculate short-term loudness (3s)",
"name": "short-term",
"value": "0x00000002"
},
{
"desc": "Calculate relative threshold and global loudness",
"name": "global",
"value": "0x00000004"
},
{
"desc": "Calculate loudness range",
"name": "loudness-range",
"value": "0x00000008"
},
{
"desc": "Calculate sample peak",
"name": "sample-peak",
"value": "0x00000010"
},
{
"desc": "Calculate true peak",
"name": "true-peak",
"value": "0x00000020"
}
]
}
},
"package": "gst-plugin-audiofx",
"source": "gst-plugin-audiofx",
"tracers": {},
"url": "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
},
"rsclosedcaption": {
Update docs
2022-10-23 17:35:14 +00:00
"description": "GStreamer Rust Closed Caption Plugin",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"elements": {
"ccdetect": {
"author": "Matthew Waters <matthew@centricular.com>",
"description": "Detect if valid closed captions are present in a stream",
"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstCCDetect",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"GstBaseTransform",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Filter/Video/ClosedCaption/Detect",
"long-name": "Closed Caption Detect",
"pad-templates": {
"sink": {
"caps": "closedcaption/x-cea-708:\n format: { cc_data, cdp }\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "closedcaption/x-cea-708:\n format: { cc_data, cdp }\n",
"direction": "src",
"presence": "always"
}
},
"properties": {
"cc608": {
"blurb": "Whether CEA608 captions (CC1/CC3) have been detected",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "null",
"readable": true,
"type": "gboolean",
"writable": false
},
"cc708": {
"blurb": "Whether CEA708 captions (cc_data) have been detected",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "null",
"readable": true,
"type": "gboolean",
"writable": false
},
"window": {
"blurb": "Window of time (in ns) to determine if captions exist in the stream",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "10000000000",
"max": "18446744073709551614",
"min": "0",
"mutable": "playing",
"readable": true,
"type": "guint64",
"writable": true
}
},
"rank": "none"
},
"cea608overlay": {
"author": "Mathieu Duponchelle <mathieu@centricular.com>",
"description": "Renders CEA 608 closed caption meta over raw video frames",
"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstCea608Overlay",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Video/Overlay/Subtitle",
"long-name": "Cea 608 overlay",
"pad-templates": {
"sink": {
Update plugins cache JSON for new CI GStreamer version Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1570>
2024-05-10 11:14:51 +00:00
"caps": "video/x-raw:\n format: { A444_16LE, A444_16BE, Y416_LE, AYUV64, RGBA64_LE, ARGB64, ARGB64_LE, BGRA64_LE, ABGR64_LE, Y416_BE, RGBA64_BE, ARGB64_BE, BGRA64_BE, ABGR64_BE, A422_16LE, A422_16BE, A420_16LE, A420_16BE, A444_12LE, GBRA_12LE, A444_12BE, GBRA_12BE, Y412_LE, Y412_BE, A422_12LE, A422_12BE, A420_12LE, A420_12BE, A444_10LE, GBRA_10LE, A444_10BE, GBRA_10BE, A422_10LE, A422_10BE, A420_10LE, A420_10BE, BGR10A2_LE, RGB10A2_LE, Y410, A444, GBRA, AYUV, VUYA, RGBA, RBGA, ARGB, BGRA, ABGR, A422, A420, AV12, Y444_16LE, GBR_16LE, Y444_16BE, GBR_16BE, Y216_LE, Y216_BE, v216, P016_LE, P016_BE, Y444_12LE, GBR_12LE, Y444_12BE, GBR_12BE, I422_12LE, I422_12BE, Y212_LE, Y212_BE, I420_12LE, I420_12BE, P012_LE, P012_BE, Y444_10LE, GBR_10LE, Y444_10BE, GBR_10BE, r210, I422_10LE, I422_10BE, NV16_10LE32, Y210, UYVP, v210, I420_10LE, I420_10BE, P010_10LE, NV12_10LE40, NV12_10LE32, P010_10BE, MT2110R, MT2110T, NV12_10BE_8L128, NV12_10LE40_4L4, Y444, BGRP, GBR, RGBP, NV24, v308, IYU2, RGBx, xRGB, BGRx, xBGR, RGB, BGR, Y42B, NV16, NV61, YUY2, YVYU, UYVY, VYUY, I420, YV12, NV12, NV21, NV12_16L32S, NV12_32L32, NV12_4L4, NV12_64Z32, NV12_8L128, Y41B, IYU1, YUV9, YVU9, BGR16, RGB16, BGR15, RGB15, RGB8P, GRAY16_LE, GRAY16_BE, GRAY10_LE32, GRAY8 }\n width: [ 1, 2147483647 ]\n height: [ 1, 2147483647 ]\n framerate: [ 0/1, 2147483647/1 ]\n",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"direction": "sink",
"presence": "always"
},
"src": {
Update plugins cache JSON for new CI GStreamer version Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1570>
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"caps": "video/x-raw:\n format: { A444_16LE, A444_16BE, Y416_LE, AYUV64, RGBA64_LE, ARGB64, ARGB64_LE, BGRA64_LE, ABGR64_LE, Y416_BE, RGBA64_BE, ARGB64_BE, BGRA64_BE, ABGR64_BE, A422_16LE, A422_16BE, A420_16LE, A420_16BE, A444_12LE, GBRA_12LE, A444_12BE, GBRA_12BE, Y412_LE, Y412_BE, A422_12LE, A422_12BE, A420_12LE, A420_12BE, A444_10LE, GBRA_10LE, A444_10BE, GBRA_10BE, A422_10LE, A422_10BE, A420_10LE, A420_10BE, BGR10A2_LE, RGB10A2_LE, Y410, A444, GBRA, AYUV, VUYA, RGBA, RBGA, ARGB, BGRA, ABGR, A422, A420, AV12, Y444_16LE, GBR_16LE, Y444_16BE, GBR_16BE, Y216_LE, Y216_BE, v216, P016_LE, P016_BE, Y444_12LE, GBR_12LE, Y444_12BE, GBR_12BE, I422_12LE, I422_12BE, Y212_LE, Y212_BE, I420_12LE, I420_12BE, P012_LE, P012_BE, Y444_10LE, GBR_10LE, Y444_10BE, GBR_10BE, r210, I422_10LE, I422_10BE, NV16_10LE32, Y210, UYVP, v210, I420_10LE, I420_10BE, P010_10LE, NV12_10LE40, NV12_10LE32, P010_10BE, MT2110R, MT2110T, NV12_10BE_8L128, NV12_10LE40_4L4, Y444, BGRP, GBR, RGBP, NV24, v308, IYU2, RGBx, xRGB, BGRx, xBGR, RGB, BGR, Y42B, NV16, NV61, YUY2, YVYU, UYVY, VYUY, I420, YV12, NV12, NV21, NV12_16L32S, NV12_32L32, NV12_4L4, NV12_64Z32, NV12_8L128, Y41B, IYU1, YUV9, YVU9, BGR16, RGB16, BGR15, RGB15, RGB8P, GRAY16_LE, GRAY16_BE, GRAY10_LE32, GRAY8 }\n width: [ 1, 2147483647 ]\n height: [ 1, 2147483647 ]\n framerate: [ 0/1, 2147483647/1 ]\n",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"direction": "src",
"presence": "always"
}
},
"properties": {
"black-background": {
"blurb": "Whether a black background should be drawn behind text",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "playing",
"readable": true,
"type": "gboolean",
"writable": true
},
"field": {
"blurb": "The field to render the caption for when available, (-1=automatic)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "-1",
"max": "1",
"min": "-1",
"mutable": "playing",
"readable": true,
"type": "gint",
"writable": true
},
"timeout": {
"blurb": "Duration after which to erase overlay when no cc data has arrived for the selected field",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "18446744073709551615",
"max": "18446744073709551615",
"min": "16000000000",
"mutable": "playing",
"readable": true,
"type": "guint64",
"writable": true
}
},
"rank": "primary"
},
closedcaption: add cea608tocea708 element Implement an element that can take an input 608 caption stream and generate a valid 708 caption stream by parsing the 608 data and generating the equivalent DTVCCPackets and Service blocks. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1112>
2023-03-01 02:06:11 +00:00
"cea608tocea708": {
"author": "Matthew Waters <matthew@centricular.com>",
"description": "Converts CEA-608 Closed Captions to CEA-708 Closed Captions",
"hierarchy": [
"GstCea608ToCea708",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Converter",
"pad-templates": {
"sink": {
"caps": "closedcaption/x-cea-608:\n format: s334-1a\nclosedcaption/x-cea-608:\n format: raw\n field: { (int)0, (int)1 }\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "closedcaption/x-cea-708:\n format: cc_data\n",
"direction": "src",
"presence": "always"
}
},
"rank": "none"
},
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"cea608tojson": {
"author": "Mathieu Duponchelle <mathieu@centricular.com>",
"description": "Converts CEA-608 Closed Captions to JSON",
"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstCea608ToJson",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Generic",
"long-name": "CEA-608 to TT",
"pad-templates": {
"sink": {
"caps": "closedcaption/x-cea-608:\n format: raw\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "application/x-json:\n",
"direction": "src",
"presence": "always"
}
},
"properties": {
"unbuffered": {
"blurb": "Whether captions should be output at display time, instead of waiting to determine durations. Useful with live input",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "ready",
"readable": true,
"type": "gboolean",
"writable": true
}
},
"rank": "none"
},
"cea608tott": {
"author": "Sebastian Dröge <sebastian@centricular.com>",
"description": "Converts CEA-608 Closed Captions to SRT/VTT timed text",
"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstCea608ToTt",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Generic",
"long-name": "CEA-608 to TT",
"pad-templates": {
"sink": {
"caps": "closedcaption/x-cea-608:\n format: raw\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "application/x-subtitle-vtt:\napplication/x-subtitle:\ntext/x-raw:\n format: utf8\n",
"direction": "src",
"presence": "always"
}
},
"rank": "none"
},
cea708mux: add element muxing multiple 708 caption services together Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1406>
2023-11-21 13:10:03 +00:00
"cea708mux": {
"author": "Matthew Waters <matthew@centricular.com>",
"description": "Combines multiple CEA-708 streams",
"hierarchy": [
"GstCea708Mux",
"GstAggregator",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Muxer",
"pad-templates": {
"sink_%%u": {
"caps": "closedcaption/x-cea-708:\n format: cc_data\n framerate: { (fraction)60/1, (fraction)60000/1001, (fraction)50/1, (fraction)30/1, (fraction)30000/1001, (fraction)25/1, (fraction)24/1, (fraction)24000/1001 }\n",
"direction": "sink",
"presence": "request",
"type": "GstCea708MuxSinkPad"
},
"src": {
"caps": "closedcaption/x-cea-708:\n format: cc_data\n framerate: { (fraction)60/1, (fraction)60000/1001, (fraction)50/1, (fraction)30/1, (fraction)30000/1001, (fraction)25/1, (fraction)24/1, (fraction)24000/1001 }\n",
"direction": "src",
"presence": "always",
"type": "GstAggregatorPad"
}
},
"rank": "none"
},
closedcaption: implement cea708overlay element Can overlay any single CEA-708 service or any single CEA-608 channel. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1519>
2024-03-25 02:25:55 +00:00
"cea708overlay": {
"author": "Matthew Waters <matthew@centricular.com>",
"description": "Renders CEA 708 closed caption meta over raw video frames",
"hierarchy": [
"GstCea708Overlay",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Video/Overlay/Subtitle",
"pad-templates": {
"sink": {
Update plugins cache JSON for new CI GStreamer version Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1570>
2024-05-10 11:14:51 +00:00
"caps": "video/x-raw:\n format: { A444_16LE, A444_16BE, Y416_LE, AYUV64, RGBA64_LE, ARGB64, ARGB64_LE, BGRA64_LE, ABGR64_LE, Y416_BE, RGBA64_BE, ARGB64_BE, BGRA64_BE, ABGR64_BE, A422_16LE, A422_16BE, A420_16LE, A420_16BE, A444_12LE, GBRA_12LE, A444_12BE, GBRA_12BE, Y412_LE, Y412_BE, A422_12LE, A422_12BE, A420_12LE, A420_12BE, A444_10LE, GBRA_10LE, A444_10BE, GBRA_10BE, A422_10LE, A422_10BE, A420_10LE, A420_10BE, BGR10A2_LE, RGB10A2_LE, Y410, A444, GBRA, AYUV, VUYA, RGBA, RBGA, ARGB, BGRA, ABGR, A422, A420, AV12, Y444_16LE, GBR_16LE, Y444_16BE, GBR_16BE, Y216_LE, Y216_BE, v216, P016_LE, P016_BE, Y444_12LE, GBR_12LE, Y444_12BE, GBR_12BE, I422_12LE, I422_12BE, Y212_LE, Y212_BE, I420_12LE, I420_12BE, P012_LE, P012_BE, Y444_10LE, GBR_10LE, Y444_10BE, GBR_10BE, r210, I422_10LE, I422_10BE, NV16_10LE32, Y210, UYVP, v210, I420_10LE, I420_10BE, P010_10LE, NV12_10LE40, NV12_10LE32, P010_10BE, MT2110R, MT2110T, NV12_10BE_8L128, NV12_10LE40_4L4, Y444, BGRP, GBR, RGBP, NV24, v308, IYU2, RGBx, xRGB, BGRx, xBGR, RGB, BGR, Y42B, NV16, NV61, YUY2, YVYU, UYVY, VYUY, I420, YV12, NV12, NV21, NV12_16L32S, NV12_32L32, NV12_4L4, NV12_64Z32, NV12_8L128, Y41B, IYU1, YUV9, YVU9, BGR16, RGB16, BGR15, RGB15, RGB8P, GRAY16_LE, GRAY16_BE, GRAY10_LE32, GRAY8 }\n width: [ 1, 2147483647 ]\n height: [ 1, 2147483647 ]\n framerate: [ 0/1, 2147483647/1 ]\n",
closedcaption: implement cea708overlay element Can overlay any single CEA-708 service or any single CEA-608 channel. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1519>
2024-03-25 02:25:55 +00:00
"direction": "sink",
"presence": "always"
},
"src": {
Update plugins cache JSON for new CI GStreamer version Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1570>
2024-05-10 11:14:51 +00:00
"caps": "video/x-raw:\n format: { A444_16LE, A444_16BE, Y416_LE, AYUV64, RGBA64_LE, ARGB64, ARGB64_LE, BGRA64_LE, ABGR64_LE, Y416_BE, RGBA64_BE, ARGB64_BE, BGRA64_BE, ABGR64_BE, A422_16LE, A422_16BE, A420_16LE, A420_16BE, A444_12LE, GBRA_12LE, A444_12BE, GBRA_12BE, Y412_LE, Y412_BE, A422_12LE, A422_12BE, A420_12LE, A420_12BE, A444_10LE, GBRA_10LE, A444_10BE, GBRA_10BE, A422_10LE, A422_10BE, A420_10LE, A420_10BE, BGR10A2_LE, RGB10A2_LE, Y410, A444, GBRA, AYUV, VUYA, RGBA, RBGA, ARGB, BGRA, ABGR, A422, A420, AV12, Y444_16LE, GBR_16LE, Y444_16BE, GBR_16BE, Y216_LE, Y216_BE, v216, P016_LE, P016_BE, Y444_12LE, GBR_12LE, Y444_12BE, GBR_12BE, I422_12LE, I422_12BE, Y212_LE, Y212_BE, I420_12LE, I420_12BE, P012_LE, P012_BE, Y444_10LE, GBR_10LE, Y444_10BE, GBR_10BE, r210, I422_10LE, I422_10BE, NV16_10LE32, Y210, UYVP, v210, I420_10LE, I420_10BE, P010_10LE, NV12_10LE40, NV12_10LE32, P010_10BE, MT2110R, MT2110T, NV12_10BE_8L128, NV12_10LE40_4L4, Y444, BGRP, GBR, RGBP, NV24, v308, IYU2, RGBx, xRGB, BGRx, xBGR, RGB, BGR, Y42B, NV16, NV61, YUY2, YVYU, UYVY, VYUY, I420, YV12, NV12, NV21, NV12_16L32S, NV12_32L32, NV12_4L4, NV12_64Z32, NV12_8L128, Y41B, IYU1, YUV9, YVU9, BGR16, RGB16, BGR15, RGB15, RGB8P, GRAY16_LE, GRAY16_BE, GRAY10_LE32, GRAY8 }\n width: [ 1, 2147483647 ]\n height: [ 1, 2147483647 ]\n framerate: [ 0/1, 2147483647/1 ]\n",
closedcaption: implement cea708overlay element Can overlay any single CEA-708 service or any single CEA-608 channel. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1519>
2024-03-25 02:25:55 +00:00
"direction": "src",
"presence": "always"
}
},
"properties": {
"cea608-channel": {
"blurb": "The cea608 channel (CC1-4) to render the caption for when available, (-1=automatic, 0=disabled)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "-1",
"max": "4",
"min": "-1",
"mutable": "playing",
"readable": true,
"type": "gint",
"writable": true
},
"service": {
"blurb": "The service to render the caption for when available, (-1=automatic, 0=disabled)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "1",
"max": "31",
"min": "-1",
"mutable": "playing",
"readable": true,
"type": "gint",
"writable": true
},
"timeout": {
"blurb": "Duration after which to erase overlay when no cc data has arrived for the selected service/channel",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "18446744073709551615",
"max": "18446744073709551615",
"min": "16000000000",
"mutable": "playing",
"readable": true,
"type": "guint64",
"writable": true
}
},
"rank": "primary"
},
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"jsontovtt": {
"author": "Jan Schmidt <jan@centricular.com>",
"description": "Converts JSON to WebVTT",
"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstJsonToVtt",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Generic",
"long-name": "JSON to WebVTT",
"pad-templates": {
"sink": {
"caps": "application/x-json:\n format: cea608\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "application/x-subtitle-vtt-fragmented:\n inline-headers: true\n",
"direction": "src",
"presence": "always"
}
},
jsontovtt: Add timeout property As described in the spec D.4 Automatic Caption Blanking, allows automatic clear if user specified timeout value
2022-10-10 13:15:57 +00:00
"properties": {
"timeout": {
"blurb": "Duration after which to erase text when no data has arrived",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "18446744073709551615",
"max": "18446744073709551615",
"min": "16000000000",
"mutable": "playing",
"readable": true,
"type": "guint64",
"writable": true
}
},
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"rank": "none"
},
"mccenc": {
"author": "Sebastian Dröge <sebastian@centricular.com>",
"description": "Encodes MCC Closed Caption Files",
"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstMccEnc",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Encoder/ClosedCaption",
"long-name": "Mcc Encoder",
"pad-templates": {
"sink": {
"caps": "closedcaption/x-cea-708:\n format: cdp\n framerate: { (fraction)24/1, (fraction)25/1, (fraction)30000/1001, (fraction)30/1, (fraction)50/1, (fraction)60000/1001, (fraction)60/1 }\nclosedcaption/x-cea-608:\n format: s334-1a\n framerate: { (fraction)24/1, (fraction)25/1, (fraction)30000/1001, (fraction)30/1, (fraction)50/1, (fraction)60000/1001, (fraction)60/1 }\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "application/x-mcc:\n",
"direction": "src",
"presence": "always"
}
},
"properties": {
"creation-date": {
"blurb": "Creation date for the output file",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"mutable": "ready",
"readable": true,
"type": "GDateTime",
"writable": true
},
"uuid": {
"blurb": "UUID for the output file",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
}
},
"rank": "primary"
},
"mccparse": {
"author": "Sebastian Dröge <sebastian@centricular.com>",
"description": "Parses MCC Closed Caption Files",
"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstMccParse",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Parser/ClosedCaption",
"long-name": "Mcc Parse",
"pad-templates": {
"sink": {
"caps": "application/x-mcc:\n version: { (int)1, (int)2 }\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "closedcaption/x-cea-708:\n format: cdp\n framerate: [ 1/2147483647, 2147483647/1 ]\nclosedcaption/x-cea-608:\n format: s334-1a\n framerate: [ 1/2147483647, 2147483647/1 ]\n",
"direction": "src",
"presence": "always"
}
},
"rank": "primary"
},
"sccenc": {
"author": "Sebastian Dröge <sebastian@centricular.com>, Jordan Petridis <jordan@centricular.com>",
"description": "Encodes SCC Closed Caption Files",
"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstSccEnc",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Encoder/ClosedCaption",
"long-name": "Scc Encoder",
"pad-templates": {
"sink": {
"caps": "closedcaption/x-cea-608:\n format: raw\n framerate: { (fraction)30000/1001, (fraction)30/1 }\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "application/x-scc:\n",
"direction": "src",
"presence": "always"
}
},
"properties": {
"output-padding": {
"blurb": "Whether the encoder should output padding captions. The element will never add padding, but will encode padding buffers it receives if this property is set to true.",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "true",
"mutable": "ready",
"readable": true,
"type": "gboolean",
"writable": true
}
},
"rank": "primary"
},
"sccparse": {
"author": "Sebastian Dröge <sebastian@centricular.com>, Jordan Petridis <jordan@centricular.com>",
"description": "Parses SCC Closed Caption Files",
"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstSccParse",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Parser/ClosedCaption",
"long-name": "Scc Parse",
"pad-templates": {
"sink": {
"caps": "application/x-scc:\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "closedcaption/x-cea-608:\n format: raw\n framerate: { (fraction)30000/1001, (fraction)30/1 }\n",
"direction": "src",
"presence": "always"
}
},
"rank": "primary"
},
"transcriberbin": {
"author": "Mathieu Duponchelle <mathieu@centricular.com>",
"description": "Transcribes audio and adds it as closed captions",
"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstTranscriberBin",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"GstBin",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"interfaces": [
"GstChildProxy"
],
"klass": "Audio / Video / Text",
"long-name": "TranscriberBin",
"pad-templates": {
"sink_audio": {
"caps": "audio/x-raw:\n",
"direction": "sink",
transcriberbin: add support for consuming secondary audio streams In some situations, a translated alternate audio stream for a content might be available. Instead of going through transcription and translation of the original audio stream, it may be preferrable for accuracy purposes to simply transcribe the secondary audio stream. This MR adds support for doing just that: * Secondary audio sink pads can be requested as "sink_audio_%u" * Sometimes audio source pads are added at that point to pass through the audio, as "src_audio_%u" * The main transcription bin now contains per-input stream transcription bins. Those can be individually controlled through properties on the sink pads, for instance translation-languages can be dynamically set per audio stream * Some properties that originally existed on the main element still remain, but are now simply mapped to the always audio sink pad * Releasing of secondary sink pads is nominally implemented, but not tested in states other than NULL An example launch line for this would be: ``` $ gst-launch-1.0 transcriberbin name=transcriberbin latency=8000 accumulate-time=0 \ cc-caps="closedcaption/x-cea-708, format=cc_data" sink_audio_0::language-code="es-US" \ sink_audio_0::translation-languages="languages, transcript=cc3" uridecodebin uri=file:///home/meh/Music/chaplin.mkv name=d d. ! videoconvert ! transcriberbin.sink_video d. ! clocksync ! audioconvert ! transcriberbin.sink_audio transcriberbin.src_video ! cea608overlay field=1 ! videoconvert ! autovideosink \ transcriberbin.src_audio ! audioconvert ! fakesink \ uridecodebin uri=file:///home/meh/Music/chaplin-spanish.webm name=d2 \ d2. ! audioconvert ! transcriberbin.sink_audio_0 \ transcriberbin.src_audio_0 ! fakesink ``` Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1546>
2024-04-19 18:12:46 +00:00
"presence": "always",
"type": "GstTranscriberSinkPad"
},
"sink_audio_%%u": {
"caps": "audio/x-raw:\n",
"direction": "sink",
"presence": "request",
"type": "GstTranscriberSinkPad"
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
},
"sink_video": {
transcriberbin: Allow video with ANY caps features transcriberbin does not read/write video buffers actually. Allow ANY caps features in order to avoid unnecessary GPU upload/download Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1165>
2023-04-07 16:51:12 +00:00
"caps": "video/x-raw(ANY):\n",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"direction": "sink",
"presence": "always"
},
"src_audio": {
"caps": "audio/x-raw:\n",
"direction": "src",
"presence": "always"
},
transcriberbin: add support for consuming secondary audio streams In some situations, a translated alternate audio stream for a content might be available. Instead of going through transcription and translation of the original audio stream, it may be preferrable for accuracy purposes to simply transcribe the secondary audio stream. This MR adds support for doing just that: * Secondary audio sink pads can be requested as "sink_audio_%u" * Sometimes audio source pads are added at that point to pass through the audio, as "src_audio_%u" * The main transcription bin now contains per-input stream transcription bins. Those can be individually controlled through properties on the sink pads, for instance translation-languages can be dynamically set per audio stream * Some properties that originally existed on the main element still remain, but are now simply mapped to the always audio sink pad * Releasing of secondary sink pads is nominally implemented, but not tested in states other than NULL An example launch line for this would be: ``` $ gst-launch-1.0 transcriberbin name=transcriberbin latency=8000 accumulate-time=0 \ cc-caps="closedcaption/x-cea-708, format=cc_data" sink_audio_0::language-code="es-US" \ sink_audio_0::translation-languages="languages, transcript=cc3" uridecodebin uri=file:///home/meh/Music/chaplin.mkv name=d d. ! videoconvert ! transcriberbin.sink_video d. ! clocksync ! audioconvert ! transcriberbin.sink_audio transcriberbin.src_video ! cea608overlay field=1 ! videoconvert ! autovideosink \ transcriberbin.src_audio ! audioconvert ! fakesink \ uridecodebin uri=file:///home/meh/Music/chaplin-spanish.webm name=d2 \ d2. ! audioconvert ! transcriberbin.sink_audio_0 \ transcriberbin.src_audio_0 ! fakesink ``` Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1546>
2024-04-19 18:12:46 +00:00
"src_audio_%%u": {
"caps": "audio/x-raw:\n",
"direction": "src",
"presence": "sometimes",
"type": "GstTranscriberSrcPad"
},
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"src_video": {
transcriberbin: Allow video with ANY caps features transcriberbin does not read/write video buffers actually. Allow ANY caps features in order to avoid unnecessary GPU upload/download Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1165>
2023-04-07 16:51:12 +00:00
"caps": "video/x-raw(ANY):\n",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"direction": "src",
"presence": "always"
}
},
"properties": {
"accumulate-time": {
"blurb": "Cut-off time for textwrap accumulation, in milliseconds (0=do not accumulate). Set this to a non-default value if you plan to switch to pop-on mode",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "-1",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint",
"writable": true
},
"caption-source": {
"blurb": "Caption source to use. If \"Transcription\" or \"Inband\" is selected, the caption meta of the other source will be dropped by transcriberbin",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "both (0)",
"mutable": "playing",
"readable": true,
"type": "GstTranscriberBinCaptionSource",
"writable": true
},
"cc-caps": {
"blurb": "The expected format of the closed captions",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "closedcaption/x-cea-608, format=(string)raw",
"mutable": "ready",
"readable": true,
"type": "GstCaps",
"writable": true
},
transcriberbin: Add "language-code" property Proxy the child transcriber element's property so that transcriberbin can apply the property with required state management Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1205>
2023-05-09 14:24:15 +00:00
"language-code": {
"blurb": "The language of the input stream",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "en-US",
"mutable": "playing",
"readable": true,
"type": "gchararray",
"writable": true
},
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"latency": {
"blurb": "Amount of milliseconds to allow the transcriber",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "4000",
"max": "-1",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint",
"writable": true
},
"mode": {
"blurb": "Which closed caption mode to operate in",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "roll-up2 (2)",
"mutable": "playing",
"readable": true,
"type": "GstTtToCea608Mode",
"writable": true
},
transcriberbin: add support for generating cea708 captions Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1406>
2023-11-21 13:09:22 +00:00
"mux-method": {
"blurb": "The method for muxing multiple transcription streams",
"conditionally-available": false,
transcriberbin: also support 608 inside 708 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1406>
2023-12-12 03:20:43 +00:00
"construct": true,
transcriberbin: add support for generating cea708 captions Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1406>
2023-11-21 13:09:22 +00:00
"construct-only": false,
"controllable": false,
"default": "cea608 (0)",
"mutable": "null",
"readable": true,
"type": "GstTranscriberBinMuxMethod",
"writable": true
},
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"passthrough": {
"blurb": "Whether transcription should occur",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "playing",
"readable": true,
"type": "gboolean",
"writable": true
},
"transcriber": {
"blurb": "The transcriber element to use",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"mutable": "ready",
"readable": true,
"type": "GstElement",
"writable": true
transcriberbin: add support for translations With this, if the transcriber element in use supports "translation_src_" request source pads, the user can now specify what languages to translate to and how to map them to 608 channels (only CC1 and CC3 are supported). For instance, translation-languages="languages, CC3=transcript, CC1=fr" will cause the original transcript to be muxed into the CC3 channel, and the French translation to be muxed into the CC1 channel. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1149>
2023-03-24 23:14:46 +00:00
},
"translate-latency": {
"blurb": "Amount of extra milliseconds to allow for translating",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "500",
"max": "-1",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint",
"writable": true
},
"translation-languages": {
transcriberbin: add support for generating cea708 captions Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1406>
2023-11-21 13:09:22 +00:00
"blurb": "A map of language codes to caption channels, e.g. translation-languages=\"languages, transcript={CC1, 708_1}, fr={708_2, CC3}\" will map the French translation to CC1/service 1 and the original transcript to CC3/service 2",
transcriberbin: add support for translations With this, if the transcriber element in use supports "translation_src_" request source pads, the user can now specify what languages to translate to and how to map them to 608 channels (only CC1 and CC3 are supported). For instance, translation-languages="languages, CC3=transcript, CC1=fr" will cause the original transcript to be muxed into the CC3 channel, and the French translation to be muxed into the CC1 channel. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1149>
2023-03-24 23:14:46 +00:00
"conditionally-available": false,
transcriberbin: Fix mux-method=cea708 * Update "translation-languages" property to include G_PARAM_CONSTRUCT so that it can be applied to initial state. * Change default "translation-languages" value to be None instead of cea608 specific one. Transcriberbin will be able to configure initia state depending on selected mux method if "translation-languages" is unspecified. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1589>
2024-05-28 12:54:54 +00:00
"construct": true,
transcriberbin: Add support for runtime translation-languages update Allows updating translation-languages at runtime Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1162>
2023-04-04 16:25:40 +00:00
"construct-only": false,
transcriberbin: add support for translations With this, if the transcriber element in use supports "translation_src_" request source pads, the user can now specify what languages to translate to and how to map them to 608 channels (only CC1 and CC3 are supported). For instance, translation-languages="languages, CC3=transcript, CC1=fr" will cause the original transcript to be muxed into the CC3 channel, and the French translation to be muxed into the CC1 channel. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1149>
2023-03-24 23:14:46 +00:00
"controllable": false,
transcriberbin: add support for consuming secondary audio streams In some situations, a translated alternate audio stream for a content might be available. Instead of going through transcription and translation of the original audio stream, it may be preferrable for accuracy purposes to simply transcribe the secondary audio stream. This MR adds support for doing just that: * Secondary audio sink pads can be requested as "sink_audio_%u" * Sometimes audio source pads are added at that point to pass through the audio, as "src_audio_%u" * The main transcription bin now contains per-input stream transcription bins. Those can be individually controlled through properties on the sink pads, for instance translation-languages can be dynamically set per audio stream * Some properties that originally existed on the main element still remain, but are now simply mapped to the always audio sink pad * Releasing of secondary sink pads is nominally implemented, but not tested in states other than NULL An example launch line for this would be: ``` $ gst-launch-1.0 transcriberbin name=transcriberbin latency=8000 accumulate-time=0 \ cc-caps="closedcaption/x-cea-708, format=cc_data" sink_audio_0::language-code="es-US" \ sink_audio_0::translation-languages="languages, transcript=cc3" uridecodebin uri=file:///home/meh/Music/chaplin.mkv name=d d. ! videoconvert ! transcriberbin.sink_video d. ! clocksync ! audioconvert ! transcriberbin.sink_audio transcriberbin.src_video ! cea608overlay field=1 ! videoconvert ! autovideosink \ transcriberbin.src_audio ! audioconvert ! fakesink \ uridecodebin uri=file:///home/meh/Music/chaplin-spanish.webm name=d2 \ d2. ! audioconvert ! transcriberbin.sink_audio_0 \ transcriberbin.src_audio_0 ! fakesink ``` Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1546>
2024-04-19 18:12:46 +00:00
"default": "languages, transcript=(string)cc1;",
transcriberbin: Add support for runtime translation-languages update Allows updating translation-languages at runtime Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1162>
2023-04-04 16:25:40 +00:00
"mutable": "playing",
transcriberbin: add support for translations With this, if the transcriber element in use supports "translation_src_" request source pads, the user can now specify what languages to translate to and how to map them to 608 channels (only CC1 and CC3 are supported). For instance, translation-languages="languages, CC3=transcript, CC1=fr" will cause the original transcript to be muxed into the CC3 channel, and the French translation to be muxed into the CC1 channel. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1149>
2023-03-24 23:14:46 +00:00
"readable": true,
"type": "GstStructure",
"writable": true
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
}
},
"rank": "none"
},
"tttocea608": {
"author": "Mathieu Duponchelle <mathieu@centricular.com>",
"description": "Converts timed text to CEA-608 Closed Captions",
"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstTtToCea608",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Generic",
"long-name": "TT to CEA-608",
"pad-templates": {
"sink": {
"caps": "text/x-raw:\napplication/x-json:\n format: cea608\n",
"direction": "sink",
"presence": "always"
},
"src": {
tttocea608: specify raw 608 field The element can only output field=0 raw 608 data. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1166>
2023-04-07 17:13:41 +00:00
"caps": "closedcaption/x-cea-608:\n format: raw\n framerate: [ 1/2147483647, 2147483647/1 ]\n field: 0\n",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"direction": "src",
"presence": "always"
}
},
"properties": {
"mode": {
"blurb": "Which mode to operate in",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "roll-up2 (2)",
"mutable": "playing",
"readable": true,
"type": "GstTtToCea608Mode",
"writable": true
},
"origin-column": {
"blurb": "Origin column",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "31",
"min": "0",
"mutable": "playing",
"readable": true,
"type": "guint",
"writable": true
},
"origin-row": {
"blurb": "Origin row, (-1=automatic)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "-1",
"max": "14",
"min": "-1",
"mutable": "playing",
"readable": true,
"type": "gint",
"writable": true
},
"roll-up-timeout": {
"blurb": "Duration after which to erase display memory in roll-up mode",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "18446744073709551615",
"max": "18446744073709551615",
"min": "0",
"mutable": "playing",
"readable": true,
"type": "guint64",
"writable": true
}
},
"rank": "none"
},
tttocea708: add element converting from text to cea708 captions Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1406>
2023-11-21 13:09:55 +00:00
"tttocea708": {
"author": "Matthew Waters <matthew@centricular.com>",
"description": "Converts timed text to CEA-708 Closed Captions",
"hierarchy": [
"GstTtToCea708",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Generic",
"pad-templates": {
"sink": {
"caps": "text/x-raw:\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "closedcaption/x-cea-708:\n format: cc_data\n framerate: [ 1/2147483647, 2147483647/1 ]\n",
"direction": "src",
"presence": "always"
}
},
"properties": {
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>
2023-12-08 05:33:16 +00:00
"cea608-channel": {
"blurb": "Write CEA 608 compatibility bytes with this channel, 0 = disabled (only 1 and 3 currently supported)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "4",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
tttocea708: add element converting from text to cea708 captions Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1406>
2023-11-21 13:09:55 +00:00
"mode": {
"blurb": "Which mode to operate in",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "roll-up (2)",
"mutable": "playing",
"readable": true,
"type": "GstTtToCea708Mode",
"writable": true
},
"origin-column": {
"blurb": "Origin column",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "31",
"min": "0",
"mutable": "playing",
"readable": true,
"type": "guint",
"writable": true
},
"origin-row": {
"blurb": "Origin row, (-1=automatic)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "-1",
"max": "14",
"min": "-1",
"mutable": "playing",
"readable": true,
"type": "gint",
"writable": true
},
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>
2023-12-08 05:33:16 +00:00
"roll-up-rows": {
"blurb": "Number of rows to use in roll up mode",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "2",
"max": "31",
"min": "0",
"mutable": "playing",
"readable": true,
"type": "guint",
"writable": true
},
tttocea708: add element converting from text to cea708 captions Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1406>
2023-11-21 13:09:55 +00:00
"roll-up-timeout": {
"blurb": "Duration after which to erase display memory in roll-up mode",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "18446744073709551615",
"max": "18446744073709551615",
"min": "0",
"mutable": "playing",
"readable": true,
"type": "guint64",
"writable": true
},
"service-number": {
"blurb": "Write DTVCC packets using this service",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "1",
"max": "63",
"min": "1",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
}
},
"rank": "none"
},
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"tttojson": {
"author": "Mathieu Duponchelle <mathieu@centricular.com>",
"description": "Encodes Timed Text to JSON",
"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstTtToJson",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Encoder/ClosedCaption",
"long-name": "Timed text to JSON encoder",
"pad-templates": {
"sink": {
"caps": "text/x-raw:\n format: utf8\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "application/x-json:\n",
"direction": "src",
"presence": "always"
}
},
"properties": {
"mode": {
"blurb": "Which mode to operate in",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "roll-up2 (2)",
"mutable": "ready",
"readable": true,
"type": "GstTtToCea608Mode",
"writable": true
}
},
"rank": "none"
}
},
"filename": "gstrsclosedcaption",
"license": "MPL",
"other-types": {
cea708mux: add element muxing multiple 708 caption services together Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1406>
2023-11-21 13:10:03 +00:00
"GstCea708MuxSinkPad": {
"hierarchy": [
"GstCea708MuxSinkPad",
"GstAggregatorPad",
"GstPad",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"kind": "object"
},
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"GstTranscriberBinCaptionSource": {
"kind": "enum",
"values": [
{
"desc": "Both",
"name": "both",
"value": "0"
},
{
"desc": "Transcription",
"name": "transcription",
"value": "1"
},
{
"desc": "Inband",
"name": "inband",
"value": "2"
}
]
},
transcriberbin: also support 608 inside 708 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1406>
2023-12-12 03:20:43 +00:00
"GstTranscriberBinMuxMethod": {
"kind": "enum",
"values": [
{
"desc": "Cea608",
"name": "cea608",
"value": "0"
},
{
"desc": "Cea708",
"name": "cea708",
"value": "1"
}
]
},
transcriberbin: add support for consuming secondary audio streams In some situations, a translated alternate audio stream for a content might be available. Instead of going through transcription and translation of the original audio stream, it may be preferrable for accuracy purposes to simply transcribe the secondary audio stream. This MR adds support for doing just that: * Secondary audio sink pads can be requested as "sink_audio_%u" * Sometimes audio source pads are added at that point to pass through the audio, as "src_audio_%u" * The main transcription bin now contains per-input stream transcription bins. Those can be individually controlled through properties on the sink pads, for instance translation-languages can be dynamically set per audio stream * Some properties that originally existed on the main element still remain, but are now simply mapped to the always audio sink pad * Releasing of secondary sink pads is nominally implemented, but not tested in states other than NULL An example launch line for this would be: ``` $ gst-launch-1.0 transcriberbin name=transcriberbin latency=8000 accumulate-time=0 \ cc-caps="closedcaption/x-cea-708, format=cc_data" sink_audio_0::language-code="es-US" \ sink_audio_0::translation-languages="languages, transcript=cc3" uridecodebin uri=file:///home/meh/Music/chaplin.mkv name=d d. ! videoconvert ! transcriberbin.sink_video d. ! clocksync ! audioconvert ! transcriberbin.sink_audio transcriberbin.src_video ! cea608overlay field=1 ! videoconvert ! autovideosink \ transcriberbin.src_audio ! audioconvert ! fakesink \ uridecodebin uri=file:///home/meh/Music/chaplin-spanish.webm name=d2 \ d2. ! audioconvert ! transcriberbin.sink_audio_0 \ transcriberbin.src_audio_0 ! fakesink ``` Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1546>
2024-04-19 18:12:46 +00:00
"GstTranscriberSinkPad": {
"hierarchy": [
"GstTranscriberSinkPad",
"GstGhostPad",
"GstProxyPad",
"GstPad",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"kind": "object",
"properties": {
"language-code": {
"blurb": "The language of the input stream",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "en-US",
"mutable": "playing",
"readable": true,
"type": "gchararray",
"writable": true
},
"mode": {
"blurb": "Which closed caption mode to operate in",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "roll-up2 (2)",
"mutable": "playing",
"readable": true,
"type": "GstTtToCea608Mode",
"writable": true
},
"transcriber": {
"blurb": "The transcriber element to use",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"mutable": "ready",
"readable": true,
"type": "GstElement",
"writable": true
},
"translation-languages": {
"blurb": "A map of language codes to caption channels, e.g. translation-languages=\"languages, transcript={CC1, 708_1}, fr={708_2, CC3}\" will map the French translation to CC1/service 1 and the original transcript to CC3/service 2",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "languages, transcript=(string)cc1;",
"mutable": "playing",
"readable": true,
"type": "GstStructure",
"writable": true
}
}
},
"GstTranscriberSrcPad": {
"hierarchy": [
"GstTranscriberSrcPad",
"GstGhostPad",
"GstProxyPad",
"GstPad",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"kind": "object"
},
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"GstTtToCea608Mode": {
"kind": "enum",
"values": [
{
"desc": "PopOn",
"name": "pop-on",
"value": "0"
},
{
"desc": "PaintOn",
"name": "paint-on",
"value": "1"
},
{
"desc": "RollUp2",
"name": "roll-up2",
"value": "2"
},
{
"desc": "RollUp3",
"name": "roll-up3",
"value": "3"
},
{
"desc": "RollUp4",
"name": "roll-up4",
"value": "4"
}
]
tttocea708: add element converting from text to cea708 captions Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1406>
2023-11-21 13:09:55 +00:00
},
"GstTtToCea708Mode": {
"kind": "enum",
"values": [
{
"desc": "PopOn",
"name": "pop-on",
"value": "0"
},
{
"desc": "PaintOn",
"name": "paint-on",
"value": "1"
},
{
"desc": "RollUp",
"name": "roll-up",
"value": "2"
}
]
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
}
},
"package": "gst-plugin-closedcaption",
"source": "gst-plugin-closedcaption",
"tracers": {},
"url": "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
},
"rsfile": {
Update docs
2022-10-23 17:35:14 +00:00
"description": "GStreamer Rust File Source/Sink Plugin",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"elements": {
"rsfilesink": {
"author": "François Laignel <fengalin@free.fr>, Luis de Bethencourt <luisbg@osg.samsung.com>",
"description": "Write stream to a file",
"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstRsFileSink",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"GstBaseSink",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"interfaces": [
"GstURIHandler"
],
"klass": "Sink/File",
"long-name": "File Sink",
"pad-templates": {
"sink": {
"caps": "ANY",
"direction": "sink",
"presence": "always"
}
},
"properties": {
"location": {
"blurb": "Location of the file to write",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
}
},
"rank": "none"
},
"rsfilesrc": {
"author": "François Laignel <fengalin@free.fr>, Sebastian Dröge <sebastian@centricular.com>",
"description": "Read stream from a file",
"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstRsFileSrc",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"GstBaseSrc",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"interfaces": [
"GstURIHandler"
],
"klass": "Source/File",
"long-name": "File Source",
"pad-templates": {
"src": {
"caps": "ANY",
"direction": "src",
"presence": "always"
}
},
"properties": {
"location": {
"blurb": "Location of the file to read from",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
}
},
"rank": "none"
}
},
"filename": "gstrsfile",
"license": "MIT/X11",
"other-types": {},
"package": "gst-plugin-file",
"source": "gst-plugin-file",
"tracers": {},
"url": "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
},
"rsflv": {
Update docs
2022-10-23 17:35:14 +00:00
"description": "GStreamer Rust FLV Plugin",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"elements": {
"rsflvdemux": {
"author": "Sebastian Dröge <sebastian@centricular.com>",
"description": "Demuxes FLV Streams",
"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstRsFlvDemux",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Codec/Demuxer",
"long-name": "FLV Demuxer",
"pad-templates": {
"audio": {
"caps": "audio/mpeg:\n mpegversion: 1\naudio/x-raw:\n rate: [ 1, 2147483647 ]\n channels: [ 1, 2147483647 ]\n layout: interleaved\n format: { U8, S16LE }\naudio/x-adpcm:\n layout: swf\naudio/x-nellymoser:\naudio/x-alaw:\naudio/x-mulaw:\naudio/mpeg:\n mpegversion: 4\n framed: true\n stream-format: raw\naudio/x-speex:\n",
"direction": "src",
"presence": "sometimes"
},
"sink": {
"caps": "video/x-flv:\n",
"direction": "sink",
"presence": "always"
},
"video": {
"caps": "video/x-flash-video:\n flvversion: 1\nvideo/x-flash-screen:\nvideo/x-vp6-flash:\nvideo/x-vp6-flash-alpha:\nvideo/x-flash-screen2:\nvideo/x-h264:\n stream-format: avc\nvideo/x-h263:\nvideo/mpeg:\n mpegversion: 4\n",
"direction": "src",
"presence": "sometimes"
}
},
"rank": "none"
}
},
"filename": "gstrsflv",
"license": "MIT/X11",
"other-types": {},
"package": "gst-plugin-flavors",
"source": "gst-plugin-flavors",
"tracers": {},
"url": "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
},
generic: expose inter plugin This new plugin exposes two elements, intersink and intersrc. These act as wormholes for data in the same process and can be used to forward data from one pipeline to another. The implementation makes use of gstreamer-utils' StreamProducer, and supports dynamically adding and removing consumers, before and after producers, and changing producer names while PLAYING, both on the sink and the src. This initial implementation comes with a small demo, and a few tests. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1257>
2023-06-21 17:55:27 +00:00
"rsinter": {
"description": "GStreamer Inter Plugin",
"elements": {
"intersink": {
"author": "Mathieu Duponchelle <mathieu@centricular.com>",
"description": "Inter Sink",
"hierarchy": [
"GstInterSink",
"GstBin",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"interfaces": [
"GstChildProxy"
],
"klass": "Generic/Sink",
"pad-templates": {
"sink": {
"caps": "ANY",
"direction": "sink",
"presence": "always"
}
},
"properties": {
"producer-name": {
"blurb": "Producer Name to use",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "playing",
"readable": true,
"type": "gchararray",
"writable": true
}
},
"rank": "none"
},
"intersrc": {
"author": "Mathieu Duponchelle <mathieu@centricular.com>",
"description": "Inter Src",
"hierarchy": [
"GstInterSrc",
"GstBin",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"interfaces": [
"GstChildProxy"
],
"klass": "Generic/Src",
"pad-templates": {
"src": {
"caps": "ANY",
"direction": "src",
"presence": "always"
}
},
"properties": {
"producer-name": {
"blurb": "Producer Name to consume from",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "playing",
"readable": true,
"type": "gchararray",
"writable": true
}
},
"rank": "none"
}
},
"filename": "gstrsinter",
"license": "MPL-2.0",
"other-types": {},
"package": "gst-plugin-inter",
"source": "gst-plugin-inter",
"tracers": {},
"url": "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
},
Update docs
2022-10-23 17:35:14 +00:00
"rsonvif": {
"description": "GStreamer Rust ONVIF Plugin",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"elements": {
onvif: Rename onvif(de)pay to rtponvifmetadata(de)pay and include the metadata specifier in the other element names too This is more descriptive and avoids any future conflicts with other kinds of ONVIF specific RTP (de)payloaders.
2022-08-25 11:16:37 +00:00
"onvifmetadatacombiner": {
"author": "Mathieu Duponchelle <mathieu@centricular.com>",
"description": "ONVIF metadata combiner",
"hierarchy": [
"GstOnvifMetadataCombiner",
"GstAggregator",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
onvifmetadatacombiner: Do not classify as Muxer It confuses `encodebin` and technically it is not really a muxer so as agreed on IRC, I am proposing to remove that classification.
2022-09-08 16:06:27 +00:00
"klass": "Video/Metadata/Combiner",
onvif: Rename onvif(de)pay to rtponvifmetadata(de)pay and include the metadata specifier in the other element names too This is more descriptive and avoids any future conflicts with other kinds of ONVIF specific RTP (de)payloaders.
2022-08-25 11:16:37 +00:00
"long-name": "ONVIF metadata combiner",
"pad-templates": {
"media": {
"caps": "ANY",
"direction": "sink",
"presence": "always",
"type": "GstAggregatorPad"
},
"meta": {
"caps": "application/x-onvif-metadata:\n parsed: true\n",
"direction": "sink",
"presence": "always",
"type": "GstAggregatorPad"
},
"src": {
"caps": "ANY",
"direction": "src",
"presence": "always"
}
},
"rank": "primary"
},
"onvifmetadataoverlay": {
"author": "Mathieu Duponchelle <mathieu@centricular.com>",
"description": "Renders ONVIF analytics meta over raw video frames",
"hierarchy": [
"GstOnvifMetadataOverlay",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Video/Overlay",
"long-name": "ONVIF Metadata overlay",
"pad-templates": {
"sink": {
Update plugins cache JSON for new CI GStreamer version Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1570>
2024-05-10 11:14:51 +00:00
"caps": "video/x-raw:\n format: { A444_16LE, A444_16BE, Y416_LE, AYUV64, RGBA64_LE, ARGB64, ARGB64_LE, BGRA64_LE, ABGR64_LE, Y416_BE, RGBA64_BE, ARGB64_BE, BGRA64_BE, ABGR64_BE, A422_16LE, A422_16BE, A420_16LE, A420_16BE, A444_12LE, GBRA_12LE, A444_12BE, GBRA_12BE, Y412_LE, Y412_BE, A422_12LE, A422_12BE, A420_12LE, A420_12BE, A444_10LE, GBRA_10LE, A444_10BE, GBRA_10BE, A422_10LE, A422_10BE, A420_10LE, A420_10BE, BGR10A2_LE, RGB10A2_LE, Y410, A444, GBRA, AYUV, VUYA, RGBA, RBGA, ARGB, BGRA, ABGR, A422, A420, AV12, Y444_16LE, GBR_16LE, Y444_16BE, GBR_16BE, Y216_LE, Y216_BE, v216, P016_LE, P016_BE, Y444_12LE, GBR_12LE, Y444_12BE, GBR_12BE, I422_12LE, I422_12BE, Y212_LE, Y212_BE, I420_12LE, I420_12BE, P012_LE, P012_BE, Y444_10LE, GBR_10LE, Y444_10BE, GBR_10BE, r210, I422_10LE, I422_10BE, NV16_10LE32, Y210, UYVP, v210, I420_10LE, I420_10BE, P010_10LE, NV12_10LE40, NV12_10LE32, P010_10BE, MT2110R, MT2110T, NV12_10BE_8L128, NV12_10LE40_4L4, Y444, BGRP, GBR, RGBP, NV24, v308, IYU2, RGBx, xRGB, BGRx, xBGR, RGB, BGR, Y42B, NV16, NV61, YUY2, YVYU, UYVY, VYUY, I420, YV12, NV12, NV21, NV12_16L32S, NV12_32L32, NV12_4L4, NV12_64Z32, NV12_8L128, Y41B, IYU1, YUV9, YVU9, BGR16, RGB16, BGR15, RGB15, RGB8P, GRAY16_LE, GRAY16_BE, GRAY10_LE32, GRAY8 }\n width: [ 1, 2147483647 ]\n height: [ 1, 2147483647 ]\n framerate: [ 0/1, 2147483647/1 ]\n",
onvif: Rename onvif(de)pay to rtponvifmetadata(de)pay and include the metadata specifier in the other element names too This is more descriptive and avoids any future conflicts with other kinds of ONVIF specific RTP (de)payloaders.
2022-08-25 11:16:37 +00:00
"direction": "sink",
"presence": "always"
},
"src": {
Update plugins cache JSON for new CI GStreamer version Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1570>
2024-05-10 11:14:51 +00:00
"caps": "video/x-raw:\n format: { A444_16LE, A444_16BE, Y416_LE, AYUV64, RGBA64_LE, ARGB64, ARGB64_LE, BGRA64_LE, ABGR64_LE, Y416_BE, RGBA64_BE, ARGB64_BE, BGRA64_BE, ABGR64_BE, A422_16LE, A422_16BE, A420_16LE, A420_16BE, A444_12LE, GBRA_12LE, A444_12BE, GBRA_12BE, Y412_LE, Y412_BE, A422_12LE, A422_12BE, A420_12LE, A420_12BE, A444_10LE, GBRA_10LE, A444_10BE, GBRA_10BE, A422_10LE, A422_10BE, A420_10LE, A420_10BE, BGR10A2_LE, RGB10A2_LE, Y410, A444, GBRA, AYUV, VUYA, RGBA, RBGA, ARGB, BGRA, ABGR, A422, A420, AV12, Y444_16LE, GBR_16LE, Y444_16BE, GBR_16BE, Y216_LE, Y216_BE, v216, P016_LE, P016_BE, Y444_12LE, GBR_12LE, Y444_12BE, GBR_12BE, I422_12LE, I422_12BE, Y212_LE, Y212_BE, I420_12LE, I420_12BE, P012_LE, P012_BE, Y444_10LE, GBR_10LE, Y444_10BE, GBR_10BE, r210, I422_10LE, I422_10BE, NV16_10LE32, Y210, UYVP, v210, I420_10LE, I420_10BE, P010_10LE, NV12_10LE40, NV12_10LE32, P010_10BE, MT2110R, MT2110T, NV12_10BE_8L128, NV12_10LE40_4L4, Y444, BGRP, GBR, RGBP, NV24, v308, IYU2, RGBx, xRGB, BGRx, xBGR, RGB, BGR, Y42B, NV16, NV61, YUY2, YVYU, UYVY, VYUY, I420, YV12, NV12, NV21, NV12_16L32S, NV12_32L32, NV12_4L4, NV12_64Z32, NV12_8L128, Y41B, IYU1, YUV9, YVU9, BGR16, RGB16, BGR15, RGB15, RGB8P, GRAY16_LE, GRAY16_BE, GRAY10_LE32, GRAY8 }\n width: [ 1, 2147483647 ]\n height: [ 1, 2147483647 ]\n framerate: [ 0/1, 2147483647/1 ]\n",
onvif: Rename onvif(de)pay to rtponvifmetadata(de)pay and include the metadata specifier in the other element names too This is more descriptive and avoids any future conflicts with other kinds of ONVIF specific RTP (de)payloaders.
2022-08-25 11:16:37 +00:00
"direction": "src",
"presence": "always"
}
},
"properties": {
"font-desc": {
"blurb": "Pango font description of font to be used for rendering",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "monospace 12",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
}
},
"rank": "primary"
},
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2022-08-25 22:30:08 +00:00
"onvifmetadataparse": {
"author": "Sebastian Dröge <sebastian@centricular.com>",
"description": "Parses ONVIF Timed XML Metadata",
"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstOnvifMetadataParse",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Metadata/Parser",
"long-name": "ONVIF Metadata Parser",
"pad-templates": {
"sink": {
"caps": "application/x-onvif-metadata:\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "application/x-onvif-metadata:\n parsed: true\n",
"direction": "src",
"presence": "always"
}
},
"properties": {
"latency": {
"blurb": "Maximum latency to introduce for reordering metadata (max=auto: 6s if unparsed input, 0s if parsed input)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "18446744073709551615",
"max": "18446744073709551615",
"min": "0",
"mutable": "ready",
"readable": true,
onvifmetadataparse: Push buffers from a separate source pad task to guarantee latency and generally improve correctness
2022-09-14 16:01:21 +00:00
"type": "guint64",
"writable": true
},
"max-lateness": {
"blurb": "Drop metadata that delayed by more than this",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "200000000",
"max": "18446744073709551615",
"min": "0",
"mutable": "ready",
"readable": true,
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"type": "guint64",
"writable": true
}
},
"rank": "none"
},
onvif: Rename onvif(de)pay to rtponvifmetadata(de)pay and include the metadata specifier in the other element names too This is more descriptive and avoids any future conflicts with other kinds of ONVIF specific RTP (de)payloaders.
2022-08-25 11:16:37 +00:00
"rtponvifmetadatadepay": {
"author": "Mathieu Duponchelle <mathieu@centricular.com>",
"description": "ONVIF metadata RTP depayloader",
"hierarchy": [
"GstOnvifMetadataDepay",
"GstRTPBaseDepayload",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Depayloader/Network/RTP",
"long-name": "ONVIF metadata RTP depayloader",
"pad-templates": {
"sink": {
"caps": "application/x-rtp:\n media: application\n payload: [ 96, 127 ]\n clock-rate: 90000\n encoding-name: VND.ONVIF.METADATA\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "application/x-onvif-metadata:\n",
"direction": "src",
"presence": "always"
}
},
"rank": "primary"
},
"rtponvifmetadatapay": {
"author": "Mathieu Duponchelle <mathieu@centricular.com>",
"description": "ONVIF metadata RTP payloader",
"hierarchy": [
"GstOnvifMetadataPay",
"GstRTPBasePayload",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
],
"klass": "Payloader/Network/RTP",
"long-name": "ONVIF metadata RTP payloader",
"pad-templates": {
"sink": {
"caps": "application/x-onvif-metadata:\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "application/x-rtp:\n media: application\n payload: [ 96, 127 ]\n clock-rate: 90000\n encoding-name: VND.ONVIF.METADATA\n",
"direction": "src",
"presence": "always"
}
},
"rank": "primary"
}
},
"filename": "gstrsonvif",
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 09:05:00 +00:00
"license": "MPL-2.0",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"other-types": {},
"package": "gst-plugin-onvif",
"source": "gst-plugin-onvif",
"tracers": {},
"url": "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
},
"rspng": {
Update docs
2022-10-23 17:35:14 +00:00
"description": "GStreamer Rust PNG encoder/decoder",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"elements": {
"rspngenc": {
"author": "Natanael Mojica <neithanmo@gmail>",
"description": "PNG encoder",
"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstRsPngEnc",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"GstVideoEncoder",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"interfaces": [
"GstPreset"
],
"klass": "Encoder/Video",
"long-name": "PNG encoder",
"pad-templates": {
"sink": {
"caps": "video/x-raw:\n format: { GRAY8, GRAY16_BE, RGB, RGBA }\n width: [ 1, 2147483647 ]\n height: [ 1, 2147483647 ]\n framerate: [ 0/1, 2147483647/1 ]\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "image/png:\n",
"direction": "src",
"presence": "always"
}
},
"properties": {
"compression-level": {
"blurb": "Selects the compression algorithm to use",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "default (0)",
"mutable": "ready",
"readable": true,
"type": "GstRsPngCompressionLevel",
"writable": true
},
"filter": {
"blurb": "Selects the filter type to applied",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "nofilter (0)",
"mutable": "ready",
"readable": true,
"type": "GstRsPngFilterType",
"writable": true
}
},
"rank": "primary"
}
},
"filename": "gstrspng",
"license": "MIT/X11",
"other-types": {
"GstRsPngCompressionLevel": {
"kind": "enum",
"values": [
{
"desc": "Default: Use the default compression level.",
"name": "default",
"value": "0"
},
{
"desc": "Fast: A fast compression algorithm.",
"name": "fast",
"value": "1"
},
{
"desc": "Best: Uses the algorithm with the best results.",
"name": "best",
"value": "2"
},
{
"desc": "Huffman: Huffman compression.",
"name": "huffman",
"value": "3"
},
{
"desc": "Rle: Rle compression.",
"name": "rle",
"value": "4"
}
]
},
"GstRsPngFilterType": {
"kind": "enum",
"values": [
{
"desc": "NoFilter: No filtering applied to the output.",
"name": "nofilter",
"value": "0"
},
{
"desc": "Sub: filter applied to each pixel.",
"name": "sub",
"value": "1"
},
{
"desc": "Up: Up filter similar to Sub.",
"name": "up",
"value": "2"
},
{
"desc": "Avg: The Average filter uses the average of the two neighboring pixels.",
"name": "avg",
"value": "3"
},
{
"desc": "Paeth: The Paeth filter computes a simple linear function of the three neighboring pixels.",
"name": "paeth",
"value": "4"
}
]
}
},
Update docs
2022-10-23 17:35:14 +00:00
"package": "gst-plugin-png",
"source": "gst-plugin-png",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"tracers": {},
"url": "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
},
Update docs
2022-10-23 17:35:14 +00:00
"rsrtp": {
"description": "GStreamer Rust RTP Plugin",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"elements": {
rtp: Add AC-3 RTP payloader/depayloader Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1586>
2023-12-19 13:16:29 +00:00
"rtpac3depay2": {
"author": "Tim-Philipp Müller <tim centricular com>",
"description": "Depayload an AC-3 Audio Stream from RTP packets (RFC 4184)",
"hierarchy": [
"GstRtpAc3Depay",
"GstRtpBaseDepay2",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Codec/Depayloader/Network/RTP",
"pad-templates": {
"sink": {
"caps": "application/x-rtp:\n media: audio\n encoding-name: AC3\n clock-rate: { (int)48000, (int)44100, (int)32000 }\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "audio/x-ac3:\n channels: [ 1, 6 ]\n rate: { (int)48000, (int)44100, (int)32000 }\n framed: true\n alignment: frame\n",
"direction": "src",
"presence": "always"
}
},
"rank": "marginal"
},
"rtpac3pay2": {
"author": "Tim-Philipp Müller <tim centricular com>",
"description": "Payload an AC-3 Audio Elementary Stream into RTP packets (RFC 4184)",
"hierarchy": [
"GstRtpAc3Pay",
"GstRtpBasePay2",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Codec/Payloader/Network/RTP",
"pad-templates": {
"sink": {
"caps": "audio/x-ac3:\n rate: { (int)48000, (int)44100, (int)32000 }\n channels: [ 1, 6 ]\n framed: true\n alignment: frame\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "application/x-rtp:\n media: audio\n encoding-name: AC3\n clock-rate: { (int)48000, (int)44100, (int)32000 }\n",
"direction": "src",
"presence": "always"
}
},
"properties": {
"aggregate-mode": {
"blurb": "Whether to send out audio frames immediately or aggregate them until a packet is full.",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "auto (-1)",
"mutable": "null",
"readable": true,
"type": "GstRtpAc3PayAggregateMode",
"writable": true
},
"max-ptime": {
"blurb": "Maximum duration of the packet data in ns (-1 = unlimited up to MTU)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "18446744073709551615",
"max": "9223372036854775807",
"min": "-1",
"mutable": "playing",
"readable": true,
"type": "gint64",
"writable": true
}
},
"rank": "marginal"
},
Update docs
2022-10-23 17:35:14 +00:00
"rtpav1depay": {
"author": "Vivienne Watermeier <vwatermeier@igalia.com>",
"description": "Depayload AV1 from RTP packets",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstRtpAv1Depay",
rtp: Port RTP AV1 payloader/depayloader to new base classes Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1472>
2023-11-20 15:02:55 +00:00
"GstRtpBaseDepay2",
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2022-08-25 22:30:08 +00:00
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
Update docs
2022-10-23 17:35:14 +00:00
"klass": "Codec/Depayloader/Network/RTP",
"long-name": "RTP AV1 Depayloader",
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"pad-templates": {
"sink": {
rtp: Port RTP AV1 payloader/depayloader to new base classes Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1472>
2023-11-20 15:02:55 +00:00
"caps": "application/x-rtp:\n media: video\n clock-rate: 90000\n encoding-name: AV1\n",
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2022-08-25 22:30:08 +00:00
"direction": "sink",
"presence": "always"
},
"src": {
rtpav1depay: Don't output full TUs but just OBUs as they come Simplifies state tracking and potentially reduces latency as it's not necessary to wait until all fragments of an OBU are received. The last OBU of a TU is marked with the marker flag to allow parsers to detect this without first seeing the beginning of the next TU. Also use a simple `Vec` for collecting complete OBUs instead of a `gst_base::Adapter` as this reduces the number of allocations. And also handle invalid packets a little bit more gracefully. Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/244 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1072>
2023-02-01 15:30:48 +00:00
"caps": "video/x-av1:\n parsed: true\n stream-format: obu-stream\n alignment: obu\n",
Update docs
2022-10-23 17:35:14 +00:00
"direction": "src",
"presence": "always"
}
},
"rank": "marginal"
},
"rtpav1pay": {
"author": "Vivienne Watermeier <vwatermeier@igalia.com>",
"description": "Payload AV1 as RTP packets",
"hierarchy": [
"GstRtpAv1Pay",
rtp: Port RTP AV1 payloader/depayloader to new base classes Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1472>
2023-11-20 15:02:55 +00:00
"GstRtpBasePay2",
Update docs
2022-10-23 17:35:14 +00:00
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Codec/Payloader/Network/RTP",
"long-name": "RTP AV1 payloader",
"pad-templates": {
"sink": {
rtpav1pay: Add support for tu/frame aligned input In this case every buffer can be sent out immediately and makes up a whole frame. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1072>
2023-02-02 17:51:44 +00:00
"caps": "video/x-av1:\n parsed: true\n stream-format: obu-stream\n alignment: { (string)tu, (string)frame, (string)obu }\n",
Update docs
2022-10-23 17:35:14 +00:00
"direction": "sink",
"presence": "always"
},
"src": {
rtp: Don't restrict payload types for payloaders WebRTC uses payload types 35-63 as dynamic payload types too to be able to place more codec variants into the SDP offer. Instead of allowing just certain payload types, completely remove any restrictions and let the user decide. There's technically nothing wrong with using any payload type, especially when using the encoding-name. Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/551 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1587>
2024-05-27 07:57:32 +00:00
"caps": "application/x-rtp:\n media: video\n clock-rate: 90000\n encoding-name: AV1\n",
Update docs
2022-10-23 17:35:14 +00:00
"direction": "src",
"presence": "always"
}
},
"rank": "marginal"
},
"rtpgccbwe": {
"author": "Thibault Saunier <tsaunier@igalia.com>",
"description": "Estimates current network bandwidth using the Google Congestion Control algorithm notifying about it through the 'bitrate' property",
"hierarchy": [
"GstRtpGCCBwE",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Network/WebRTC/RTP/Filter",
"long-name": "Google Congestion Control bandwidth estimator",
"pad-templates": {
"sink": {
"caps": "application/x-rtp:\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "application/x-rtp:\n",
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"direction": "src",
"presence": "always"
}
},
"properties": {
Update docs
2022-10-23 17:35:14 +00:00
"estimated-bitrate": {
"blurb": "Currently estimated bitrate. Can be set before starting\n the element to configure the starting bitrate, in which case the\n encoder should also use it as target bitrate",
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"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
Update docs
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"default": "2048000",
"max": "-1",
"min": "1",
"mutable": "ready",
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"readable": true,
Update docs
2022-10-23 17:35:14 +00:00
"type": "guint",
"writable": true
},
rtpgccbwe: Support linear regression based delay estimation In our tests, the slope (found with linear regression) on a history of the (smoothed) accumulated inter-group delays gives a more stable congestion control. In particular, low-end devices becomes less sensitive to spikes in inter-group delay measurements. This flavour of delay based bandwidth estimation with Google Congestion Control is also what Chromium is using. To make it easy to experiment with the new estimator, as well as add support for new ones in the future, also add infrastructure for making delay estimator flavour selectable at runtime. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1566>
2024-05-02 12:13:10 +00:00
"estimator": {
"blurb": "How to calculate the delay estimate that will be compared against the dynamic delay threshold.",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "kalman (0)",
"mutable": "ready",
"readable": true,
"type": "GstRtpGCCBwEEstimator",
"writable": true
},
Update docs
2022-10-23 17:35:14 +00:00
"max-bitrate": {
chore(webrtcsink): fix max-bitrate blurb and nick Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1150>
2023-03-28 05:11:05 +00:00
"blurb": "Maximum bitrate to use (in bit/sec) when computing it through the bandwidth estimation algorithm",
Update docs
2022-10-23 17:35:14 +00:00
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "8192000",
"max": "-1",
"min": "1",
"mutable": "ready",
"readable": true,
"type": "guint",
"writable": true
},
"min-bitrate": {
"blurb": "Minimal bitrate to use (in bit/sec) when computing it through the bandwidth estimation algorithm",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "1000",
"max": "-1",
"min": "1",
"mutable": "ready",
"readable": true,
"type": "guint",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"writable": true
}
},
"rank": "none"
rtp: Add PCMU/PCMA RTP payloader / depayloader elements These come with new generic RTP payloader, RTP raw-ish audio payloader and RTP depayloader base classes. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1424>
2023-10-14 08:23:56 +00:00
},
rtp: Add JPEG RTP payloader/depayloader Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1543>
2023-12-20 18:00:09 +00:00
"rtpjpegdepay2": {
"author": "Sebastian Dröge <sebastian@centricular.com>",
"description": "Depayload a JPEG Video stream from RTP packets (RFC 2435)",
"hierarchy": [
"GstRtpJpegDepay2",
"GstRtpBaseDepay2",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Codec/Depayloader/Network/RTP",
"pad-templates": {
"sink": {
"caps": "application/x-rtp:\n media: video\n payload: 26\n clock-rate: 90000\napplication/x-rtp:\n media: video\n encoding-name: JPEG\n clock-rate: 90000\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "image/jpeg:\n",
"direction": "src",
"presence": "always"
}
},
"rank": "marginal"
},
"rtpjpegpay2": {
"author": "Sebastian Dröge <sebastian@centricular.com>",
"description": "Payload a JPEG Video stream to RTP packets (RFC 2435)",
"hierarchy": [
"GstRtpJpegPay2",
"GstRtpBasePay2",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Codec/Payloader/Network/RTP",
"pad-templates": {
"sink": {
"caps": "image/jpeg:\n parsed: true\n width: [ 1, 65535 ]\n height: [ 1, 65535 ]\n sof-marker: 0\n colorspace: sYUV\n sampling: { (string)YCbCr-4:2:0, (string)YCbCr-4:2:2 }\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "application/x-rtp:\n media: video\n payload: 26\n clock-rate: 90000\napplication/x-rtp:\n media: video\n encoding-name: JPEG\n clock-rate: 90000\n",
"direction": "src",
"presence": "always"
}
},
"rank": "marginal"
},
rtp: Add KLV RTP payloader/depayloader Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1580>
2023-11-10 12:25:02 +00:00
"rtpklvdepay2": {
"author": "Tim-Philipp Müller <tim centricular com>",
"description": "Depayload an SMPTE ST 336 KLV metadata stream from RTP packets (RFC 6597)",
"hierarchy": [
"GstRtpKlvDepay2",
"GstRtpBaseDepay2",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Codec/Depayloader/Network/RTP",
"pad-templates": {
"sink": {
"caps": "application/x-rtp:\n media: application\n clock-rate: [ 1, 2147483647 ]\n encoding-name: SMPTE336M\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "meta/x-klv:\n parsed: true\n",
"direction": "src",
"presence": "always"
}
},
"rank": "marginal"
},
"rtpklvpay2": {
"author": "Tim-Philipp Müller <tim centricular com>",
"description": "Payload an SMPTE ST 336 KLV metadata stream into RTP packets (RFC 6597)",
"hierarchy": [
"GstRtpKlvPay2",
"GstRtpBasePay2",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Codec/Payloader/Network/RTP",
"pad-templates": {
"sink": {
"caps": "meta/x-klv:\n parsed: true\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "application/x-rtp:\n media: application\n encoding-name: SMPTE336M\n clock-rate: 90000\n",
"direction": "src",
"presence": "always"
}
},
"rank": "marginal"
},
rtp: Add MPEG-TS RTP depayloader Can handle different packet sizes, also see: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1310 Has clock-rate=90000 as spec prescribes, see: https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/691 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1493>
2023-10-21 15:55:03 +00:00
"rtpmp2tdepay2": {
"author": "Tim-Philipp Müller <tim centricular com>",
"description": "Depayload an MPEG Transport Stream from RTP packets (RFC 2250)",
"hierarchy": [
"GstRtpMP2TDepay2",
"GstRtpBaseDepay2",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Codec/Depayloader/Network/RTP",
"pad-templates": {
"sink": {
"caps": "application/x-rtp:\n media: video\n clock-rate: 90000\n encoding-name: MP2T\napplication/x-rtp:\n media: video\n payload: 33\n clock-rate: 90000\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "video/mpegts:\n packetsize: { (int)188, (int)192, (int)204, (int)208 }\n systemstream: true\n",
"direction": "src",
"presence": "always"
}
},
"properties": {
"skip-first-bytes": {
"blurb": "Number of bytes to skip at the beginning of the payload",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "-1",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint",
"writable": true
}
},
"rank": "marginal"
},
rtp: Add MPEG-TS RTP payloader Pushes out pending TS packets on EOS. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1493>
2023-11-21 10:19:52 +00:00
"rtpmp2tpay2": {
"author": "Tim-Philipp Müller <tim centricular com>",
"description": "Payload an MPEG Transport Stream into RTP packets (RFC 2250)",
"hierarchy": [
"GstRtpMP2TPay2",
"GstRtpBasePay2",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Codec/Payloader/Network/RTP",
"pad-templates": {
"sink": {
"caps": "video/mpegts:\n packetsize: { (int)188, (int)192, (int)204, (int)208 }\n systemstream: true\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "application/x-rtp:\n media: video\n clock-rate: 90000\n encoding-name: MP2T\napplication/x-rtp:\n media: video\n payload: 33\n clock-rate: 90000\n",
"direction": "src",
"presence": "always"
}
},
"rank": "marginal"
},
rtp: add mp4adepay Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1551>
2024-04-24 10:40:21 +00:00
"rtpmp4adepay2": {
"author": "François Laignel <francois centricular com>",
"description": "Depayload an MPEG-4 Audio bitstream (e.g. AAC) from RTP packets (RFC 3016)",
"hierarchy": [
"GstRtpMpeg4AudioDepay",
"GstRtpBaseDepay2",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Codec/Depayloader/Network/RTP",
"pad-templates": {
"sink": {
"caps": "application/x-rtp:\n media: audio\n clock-rate: [ 1, 2147483647 ]\n encoding-name: MP4A-LATM\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "audio/mpeg:\n mpegversion: 4\n framed: true\n stream-format: raw\n",
"direction": "src",
"presence": "always"
}
},
"rank": "marginal"
},
rtp: add mp4apay Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1551>
2024-04-24 12:29:14 +00:00
"rtpmp4apay2": {
"author": "François Laignel <francois centricular com>",
"description": "Payload an MPEG-4 Audio bitstream (e.g. AAC) into RTP packets (RFC 3016)",
"hierarchy": [
"GstRtpMpeg4AudioPay",
"GstRtpBasePay2",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Codec/Payloader/Network/RTP",
"pad-templates": {
"sink": {
"caps": "audio/mpeg:\n mpegversion: 4\n framed: true\n stream-format: raw\n",
"direction": "sink",
"presence": "always"
},
"src": {
rtp: Don't restrict payload types for payloaders WebRTC uses payload types 35-63 as dynamic payload types too to be able to place more codec variants into the SDP offer. Instead of allowing just certain payload types, completely remove any restrictions and let the user decide. There's technically nothing wrong with using any payload type, especially when using the encoding-name. Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/551 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1587>
2024-05-27 07:57:32 +00:00
"caps": "application/x-rtp:\n media: audio\n clock-rate: [ 1, 2147483647 ]\n encoding-name: MP4A-LATM\n",
rtp: add mp4apay Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1551>
2024-04-24 12:29:14 +00:00
"direction": "src",
"presence": "always"
}
},
"rank": "marginal"
},
rtp: add mp4gdepay Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1551>
2024-04-24 13:49:49 +00:00
"rtpmp4gdepay2": {
"author": "François Laignel <francois centricular com>",
"description": "Depayload MPEG-4 Generic elementary streams from RTP packets (RFC 3640)",
"hierarchy": [
"GstRtpMpeg4GenericDepay",
"GstRtpBaseDepay2",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Codec/Depayloader/Network/RTP",
"pad-templates": {
"sink": {
"caps": "application/x-rtp:\n media: { (string)audio, (string)video }\n clock-rate: [ 1, 2147483647 ]\n encoding-name: MPEG4-GENERIC\n mode: { (string)generic, (string)AAC-lbr, (string)AAC-hbr, (string)aac-hbr }\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "video/mpeg:\n mpegversion: 4\n systemstream: false\naudio/mpeg:\n mpegversion: 4\n stream-format: raw\n",
"direction": "src",
"presence": "always"
}
},
"rank": "marginal"
},
rtp: add mp4gpay Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1551>
2024-04-24 16:33:12 +00:00
"rtpmp4gpay2": {
"author": "François Laignel <francois centricular com>",
"description": "Payload an MPEG-4 Generic elementary stream into RTP packets (RFC 3640)",
"hierarchy": [
"GstRtpMpeg4GenericPay",
"GstRtpBasePay2",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Codec/Payloader/Network/RTP",
"pad-templates": {
"sink": {
"caps": "video/mpeg:\n mpegversion: 4\n systemstream: false\naudio/mpeg:\n mpegversion: 4\n stream-format: raw\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "application/x-rtp:\n media: { (string)audio, (string)video }\n clock-rate: [ 1, 2147483647 ]\n encoding-name: MPEG4-GENERIC\n streamtype: { (string)4, (string)5 }\n mode: { (string)generic, (string)AAC-lbr, (string)AAC-hbr, (string)aac-hbr }\n",
"direction": "src",
"presence": "always"
}
},
"properties": {
"aggregate-mode": {
"blurb": "Whether to send out AUs immediately or aggregate them until a packet is full.",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "auto (-1)",
"mutable": "null",
"readable": true,
"type": "GstRtpMpeg4GenericPayAggregateMode",
"writable": true
},
"max-ptime": {
"blurb": "Maximum duration of the packet data in ns (-1 = unlimited up to MTU)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "18446744073709551615",
"max": "9223372036854775807",
"min": "-1",
"mutable": "playing",
"readable": true,
"type": "gint64",
"writable": true
}
},
"rank": "marginal"
},
rtp: Add Opus RTP payloader/depayloader Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1571>
2023-11-01 20:13:50 +00:00
"rtpopusdepay2": {
"author": "Tim-Philipp Müller <tim centricular com>",
"description": "Depayload an Opus audio stream from RTP packets (RFC 7587)",
"hierarchy": [
"GstRtpOpusDepay2",
"GstRtpBaseDepay2",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Codec/Depayloader/Network/RTP",
"pad-templates": {
"sink": {
"caps": "application/x-rtp:\n media: audio\n encoding-name: { (string)OPUS, (string)MULTIOPUS }\n clock-rate: 48000\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "audio/x-opus:\nchannel-mapping-family: [ 0, 1 ]\n",
"direction": "src",
"presence": "always"
}
},
"rank": "marginal"
},
"rtpopuspay2": {
"author": "Tim-Philipp Müller <tim centricular com>",
"description": "Payload an Opus audio stream into RTP packets (RFC 7587)",
"hierarchy": [
"GstRtpOpusPay2",
"GstRtpBasePay2",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Codec/Payloader/Network/RTP",
"pad-templates": {
"sink": {
"caps": "audio/x-opus:\nchannel-mapping-family: 0\naudio/x-opus:\nchannel-mapping-family: 0\n channels: [ 1, 2 ]\naudio/x-opus:\nchannel-mapping-family: 1\n channels: [ 3, 255 ]\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "application/x-rtp:\n media: audio\n encoding-name: { (string)OPUS, (string)MULTIOPUS }\n clock-rate: 48000\n",
"direction": "src",
"presence": "always"
}
},
"properties": {
"dtx": {
"blurb": "Do not send out empty packets for transmission (requires opusenc dtx=true)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "playing",
"readable": true,
"type": "gboolean",
"writable": true
}
},
"rank": "marginal"
},
rtp: Add PCMU/PCMA RTP payloader / depayloader elements These come with new generic RTP payloader, RTP raw-ish audio payloader and RTP depayloader base classes. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1424>
2023-10-14 08:23:56 +00:00
"rtppcmadepay2": {
"author": "Sebastian Dröge <sebastian@centricular.com>",
"description": "Depayload A-law from RTP packets (RFC 3551)",
"hierarchy": [
"GstRtpPcmaDepay2",
"GstRtpPcmauDepay2",
"GstRtpBaseDepay2",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Codec/Depayloader/Network/RTP",
"pad-templates": {
"sink": {
"caps": "application/x-rtp:\n media: audio\n payload: 8\n clock-rate: 8000\napplication/x-rtp:\n media: audio\n clock-rate: [ 1, 2147483647 ]\n encoding-name: PCMA\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "audio/x-alaw:\n channels: 1\n rate: [ 1, 2147483647 ]\n",
"direction": "src",
"presence": "always"
}
},
"rank": "marginal"
},
"rtppcmapay2": {
"author": "Sebastian Dröge <sebastian@centricular.com>",
"description": "Payload A-law Audio into RTP packets (RFC 3551)",
"hierarchy": [
"GstRtpPcmaPay2",
"GstRtpPcmauPay2",
"GstRtpBaseAudioPay2",
"GstRtpBasePay2",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Codec/Payloader/Network/RTP",
"pad-templates": {
"sink": {
"caps": "audio/x-alaw:\n channels: 1\n rate: [ 1, 2147483647 ]\n",
"direction": "sink",
"presence": "always"
},
"src": {
rtp: Don't restrict payload types for payloaders WebRTC uses payload types 35-63 as dynamic payload types too to be able to place more codec variants into the SDP offer. Instead of allowing just certain payload types, completely remove any restrictions and let the user decide. There's technically nothing wrong with using any payload type, especially when using the encoding-name. Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/551 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1587>
2024-05-27 07:57:32 +00:00
"caps": "application/x-rtp:\n media: audio\n payload: 8\n clock-rate: 8000\napplication/x-rtp:\n media: audio\n encoding-name: PCMA\n clock-rate: [ 1, 2147483647 ]\n",
rtp: Add PCMU/PCMA RTP payloader / depayloader elements These come with new generic RTP payloader, RTP raw-ish audio payloader and RTP depayloader base classes. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1424>
2023-10-14 08:23:56 +00:00
"direction": "src",
"presence": "always"
}
},
"rank": "marginal"
},
"rtppcmudepay2": {
"author": "Sebastian Dröge <sebastian@centricular.com>",
"description": "Depayload µ-law from RTP packets (RFC 3551)",
"hierarchy": [
"GstRtpPcmuDepay2",
"GstRtpPcmauDepay2",
"GstRtpBaseDepay2",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Codec/Depayloader/Network/RTP",
"pad-templates": {
"sink": {
"caps": "application/x-rtp:\n media: audio\n payload: 0\n clock-rate: 8000\napplication/x-rtp:\n media: audio\n clock-rate: [ 1, 2147483647 ]\n encoding-name: PCMU\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "audio/x-mulaw:\n channels: 1\n rate: [ 1, 2147483647 ]\n",
"direction": "src",
"presence": "always"
}
},
"rank": "marginal"
},
"rtppcmupay2": {
"author": "Sebastian Dröge <sebastian@centricular.com>",
"description": "Payload µ-law Audio into RTP packets (RFC 3551)",
"hierarchy": [
"GstRtpPcmuPay2",
"GstRtpPcmauPay2",
"GstRtpBaseAudioPay2",
"GstRtpBasePay2",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Codec/Payloader/Network/RTP",
"pad-templates": {
"sink": {
"caps": "audio/x-mulaw:\n channels: 1\n rate: [ 1, 2147483647 ]\n",
"direction": "sink",
"presence": "always"
},
"src": {
rtp: Don't restrict payload types for payloaders WebRTC uses payload types 35-63 as dynamic payload types too to be able to place more codec variants into the SDP offer. Instead of allowing just certain payload types, completely remove any restrictions and let the user decide. There's technically nothing wrong with using any payload type, especially when using the encoding-name. Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/551 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1587>
2024-05-27 07:57:32 +00:00
"caps": "application/x-rtp:\n media: audio\n payload: 0\n clock-rate: 8000\napplication/x-rtp:\n media: audio\n encoding-name: PCMU\n clock-rate: [ 1, 2147483647 ]\n",
rtp: Add PCMU/PCMA RTP payloader / depayloader elements These come with new generic RTP payloader, RTP raw-ish audio payloader and RTP depayloader base classes. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1424>
2023-10-14 08:23:56 +00:00
"direction": "src",
"presence": "always"
}
},
"rank": "marginal"
rtp: Add VP8/9 RTP payloader/depayloader Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1487>
2023-12-12 12:53:47 +00:00
},
rtpbin2: split send and receive halves into separate elements There is now two elements, rtpsend and rtprecv that represent the two halves of a rtpsession. This avoids the potential pipeline loop if two peers are sending/receiving data towards each other. The two halves can be connected by setting the rtp-id property on each element to the same value and they will behave like a combined rtpbin-like element. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
2024-02-28 02:02:37 +00:00
"rtprecv": {
"author": "Matthew Waters <matthew@centricular.com>",
"description": "RTP sessions management (receiver)",
"hierarchy": [
"GstRtpRecv",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Network/RTP/Filter",
"pad-templates": {
"rtcp_sink_%%u": {
"caps": "application/x-rtcp:\n",
"direction": "sink",
"presence": "request"
},
"rtp_sink_%%u": {
"caps": "application/x-rtp:\n",
"direction": "sink",
"presence": "request"
},
"rtp_src_%%u_%%u_%%u": {
"caps": "application/x-rtp:\n",
"direction": "src",
"presence": "sometimes"
}
},
"properties": {
"latency": {
"blurb": "Amount of ms to buffer",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "200",
"max": "-1",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint",
"writable": true
},
"rtp-id": {
"blurb": "A connection ID shared with a rtpsend element for implementing both sending and receiving using the same RTP context",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "rtp-id",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
"stats": {
"blurb": "Statistics about the session",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"mutable": "null",
"readable": true,
"type": "guint",
"writable": false
},
"timestamping-mode": {
"blurb": "Govern how to pick presentation timestamps for packets",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "skew (2)",
"mutable": "ready",
"readable": true,
"type": "GstRtpBin2TimestampingMode",
"writable": true
}
},
"rank": "none",
"signals": {
"get-session": {
"action": true,
"args": [
{
"name": "arg0",
"type": "guint"
}
],
"return-type": "GstRtp2Session",
"when": "last"
}
}
},
"rtpsend": {
"author": "Matthew Waters <matthew@centricular.com>",
"description": "RTP session management (sender)",
"hierarchy": [
"GstRtpSend",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Network/RTP/Filter",
"pad-templates": {
"rtcp_src_%%u": {
"caps": "application/x-rtcp:\n",
"direction": "src",
"presence": "request"
},
"rtp_sink_%%u": {
"caps": "application/x-rtp:\n",
"direction": "sink",
"presence": "request"
},
"rtp_src_%%u": {
"caps": "application/x-rtp:\n",
"direction": "src",
"presence": "sometimes"
}
},
"properties": {
"min-rtcp-interval": {
"blurb": "Minimum time (in ms) between RTCP reports",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "5000",
"max": "-1",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint",
"writable": true
},
"reduced-size-rtcp": {
"blurb": "Use reduced size RTCP. Only has an effect if rtp-profile=avpf",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "ready",
"readable": true,
"type": "gboolean",
"writable": true
},
"rtp-id": {
"blurb": "A connection ID shared with a rtprecv element for implementing both sending and receiving using the same RTP context",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "rtp-id",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
"rtp-profile": {
"blurb": "RTP Profile to use",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "avp (0)",
"mutable": "ready",
"readable": true,
"type": "GstRtpSendProfile",
"writable": true
},
"stats": {
"blurb": "Statistics about the session",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"mutable": "null",
"readable": true,
"type": "guint",
"writable": false
}
},
"rank": "none",
"signals": {
"get-session": {
"action": true,
"args": [
{
"name": "arg0",
"type": "guint"
}
],
"return-type": "GstRtp2Session",
"when": "last"
}
}
},
rtp: Add VP8/9 RTP payloader/depayloader Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1487>
2023-12-12 12:53:47 +00:00
"rtpvp8depay2": {
"author": "Sebastian Dröge <sebastian@centricular.com>",
"description": "Depayload VP8 from RTP packets",
"hierarchy": [
"GstRtpVp8Depay2",
"GstRtpBaseDepay2",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Codec/Depayloader/Network/RTP",
"pad-templates": {
"sink": {
"caps": "application/x-rtp:\n media: video\n clock-rate: 90000\n encoding-name: { (string)VP8, (string)VP8-DRAFT-IETF-01 }\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "video/x-vp8:\n",
"direction": "src",
"presence": "always"
}
},
"properties": {
"request-keyframe": {
"blurb": "Request new keyframe when packet loss is detected",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "ready",
"readable": true,
"type": "gboolean",
"writable": true
},
"wait-for-keyframe": {
"blurb": "Wait for the next keyframe after packet loss",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "ready",
"readable": true,
"type": "gboolean",
"writable": true
}
},
"rank": "marginal"
},
"rtpvp8pay2": {
"author": "Sebastian Dröge <sebastian@centricular.com>",
"description": "Payload VP8 as RTP packets",
"hierarchy": [
"GstRtpVp8Pay2",
"GstRtpBasePay2",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Codec/Payloader/Network/RTP",
"pad-templates": {
"sink": {
"caps": "video/x-vp8:\n",
"direction": "sink",
"presence": "always"
},
"src": {
rtp: Don't restrict payload types for payloaders WebRTC uses payload types 35-63 as dynamic payload types too to be able to place more codec variants into the SDP offer. Instead of allowing just certain payload types, completely remove any restrictions and let the user decide. There's technically nothing wrong with using any payload type, especially when using the encoding-name. Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/551 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1587>
2024-05-27 07:57:32 +00:00
"caps": "application/x-rtp:\n media: video\n clock-rate: 90000\n encoding-name: { (string)VP8, (string)VP8-DRAFT-IETF-01 }\n",
rtp: Add VP8/9 RTP payloader/depayloader Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1487>
2023-12-12 12:53:47 +00:00
"direction": "src",
"presence": "always"
}
},
"properties": {
"fragmentation-mode": {
"blurb": "Fragmentation Mode",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "none (0)",
"mutable": "ready",
"readable": true,
"type": "GstRtpVp8Pay2FragmentationMode",
"writable": true
},
"picture-id": {
"blurb": "Current Picture ID",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "-1",
"max": "32767",
"min": "-1",
"mutable": "null",
"readable": true,
"type": "gint",
"writable": false
},
"picture-id-mode": {
"blurb": "The picture ID mode for payloading",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "none (0)",
"mutable": "ready",
"readable": true,
"type": "GstRtpVp8Pay2PictureIdMode",
"writable": true
},
"picture-id-offset": {
"blurb": "Offset to add to the initial picture-id (-1 = random)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "-1",
"max": "32767",
"min": "-1",
"mutable": "ready",
"readable": true,
"type": "gint",
"writable": true
}
},
"rank": "marginal"
},
"rtpvp9depay2": {
"author": "Sebastian Dröge <sebastian@centricular.com>",
"description": "Depayload VP9 from RTP packets",
"hierarchy": [
"GstRtpVp9Depay2",
"GstRtpBaseDepay2",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Codec/Depayloader/Network/RTP",
"pad-templates": {
"sink": {
"caps": "application/x-rtp:\n media: video\n clock-rate: 90000\n encoding-name: { (string)VP9, (string)VP9-DRAFT-IETF-01 }\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "video/x-vp9:\n",
"direction": "src",
"presence": "always"
}
},
"properties": {
"request-keyframe": {
"blurb": "Request new keyframe when packet loss is detected",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "ready",
"readable": true,
"type": "gboolean",
"writable": true
},
"wait-for-keyframe": {
"blurb": "Wait for the next keyframe after packet loss",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "ready",
"readable": true,
"type": "gboolean",
"writable": true
}
},
"rank": "marginal"
},
"rtpvp9pay2": {
"author": "Sebastian Dröge <sebastian@centricular.com>",
"description": "Payload VP9 as RTP packets",
"hierarchy": [
"GstRtpVp9Pay2",
"GstRtpBasePay2",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Codec/Payloader/Network/RTP",
"pad-templates": {
"sink": {
"caps": "video/x-vp9:\n",
"direction": "sink",
"presence": "always"
},
"src": {
rtp: Don't restrict payload types for payloaders WebRTC uses payload types 35-63 as dynamic payload types too to be able to place more codec variants into the SDP offer. Instead of allowing just certain payload types, completely remove any restrictions and let the user decide. There's technically nothing wrong with using any payload type, especially when using the encoding-name. Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/551 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1587>
2024-05-27 07:57:32 +00:00
"caps": "application/x-rtp:\n media: video\n clock-rate: 90000\n encoding-name: { (string)VP9, (string)VP9-DRAFT-IETF-01 }\n",
rtp: Add VP8/9 RTP payloader/depayloader Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1487>
2023-12-12 12:53:47 +00:00
"direction": "src",
"presence": "always"
}
},
"properties": {
"picture-id": {
"blurb": "Current Picture ID",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "-1",
"max": "32767",
"min": "-1",
"mutable": "null",
"readable": true,
"type": "gint",
"writable": false
},
"picture-id-mode": {
"blurb": "The picture ID mode for payloading",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "none (0)",
"mutable": "ready",
"readable": true,
"type": "GstRtpVp9Pay2PictureIdMode",
"writable": true
},
"picture-id-offset": {
"blurb": "Offset to add to the initial picture-id (-1 = random)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "-1",
"max": "32767",
"min": "-1",
"mutable": "ready",
"readable": true,
"type": "gint",
"writable": true
}
},
"rank": "marginal"
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
}
},
Update docs
2022-10-23 17:35:14 +00:00
"filename": "gstrsrtp",
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 09:05:00 +00:00
"license": "MPL-2.0",
rtp: Add PCMU/PCMA RTP payloader / depayloader elements These come with new generic RTP payloader, RTP raw-ish audio payloader and RTP depayloader base classes. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1424>
2023-10-14 08:23:56 +00:00
"other-types": {
rtpbin2: implement a session configuration object Currently only contains pt-map Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
2024-01-17 10:20:35 +00:00
"GstRtp2Session": {
"hierarchy": [
"GstRtp2Session",
"GObject"
],
"kind": "object",
"properties": {
"pt-map": {
"blurb": "Mapping of RTP payload type to caps",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "application/x-rtp2-pt-map;",
"mutable": "null",
"readable": true,
"type": "GstStructure",
"writable": true
}
},
"signals": {
"bye-ssrc": {
"args": [
{
"name": "arg0",
"type": "guint"
}
],
"return-type": "void",
"when": "last"
},
"new-ssrc": {
"args": [
{
"name": "arg0",
"type": "guint"
}
],
"return-type": "void",
"when": "last"
}
}
},
rtp: Add AC-3 RTP payloader/depayloader Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1586>
2023-12-19 13:16:29 +00:00
"GstRtpAc3PayAggregateMode": {
"kind": "enum",
"values": [
{
"desc": "Automatic: zero-latency if upstream is live, otherwise aggregate frames until packet is full.",
"name": "auto",
"value": "-1"
},
{
"desc": "Zero Latency: always send out frames right away, do not wait for more frames to fill a packet.",
"name": "zero-latency",
"value": "0"
},
{
"desc": "Aggregate: collect audio frames until we have a full packet or the max-ptime limit is hit (if set).",
"name": "aggregate",
"value": "1"
}
]
},
rtp: Add PCMU/PCMA RTP payloader / depayloader elements These come with new generic RTP payloader, RTP raw-ish audio payloader and RTP depayloader base classes. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1424>
2023-10-14 08:23:56 +00:00
"GstRtpBaseAudioPay2": {
"hierarchy": [
"GstRtpBaseAudioPay2",
"GstRtpBasePay2",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"kind": "object",
"properties": {
"alignment-threshold": {
"blurb": "Timestamp alignment threshold in nanoseconds",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "40000000",
"max": "18446744073709551615",
"min": "0",
"mutable": "playing",
"readable": true,
"type": "guint64",
"writable": true
},
"discont-wait": {
"blurb": "Window of time in nanoseconds to wait before creating a discontinuity",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "1000000000",
"max": "18446744073709551615",
"min": "0",
"mutable": "playing",
"readable": true,
"type": "guint64",
"writable": true
},
"max-ptime": {
"blurb": "Maximum duration of the packet data in ns (-1 = unlimited up to MTU)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "18446744073709551615",
"max": "9223372036854775807",
"min": "-1",
"mutable": "playing",
"readable": true,
"type": "gint64",
"writable": true
},
"min-ptime": {
"blurb": "Minimum duration of the packet data in ns (can't go above MTU)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "9223372036854775807",
"min": "0",
"mutable": "playing",
"readable": true,
"type": "gint64",
"writable": true
},
"ptime-multiple": {
"blurb": "Force buffers to be multiples of this duration in ns (0 disables)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "9223372036854775807",
"min": "0",
"mutable": "playing",
"readable": true,
"type": "gint64",
"writable": true
}
}
},
"GstRtpBaseDepay2": {
"hierarchy": [
"GstRtpBaseDepay2",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"kind": "object",
"properties": {
rtp2: Fix typo on auto-header-extension property name The rtp (de)pay elements use auto-header-extension so the new elements should do the same. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1646>
2024-07-02 08:35:39 +00:00
"auto-header-extension": {
rtp: Add PCMU/PCMA RTP payloader / depayloader elements These come with new generic RTP payloader, RTP raw-ish audio payloader and RTP depayloader base classes. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1424>
2023-10-14 08:23:56 +00:00
"blurb": "Whether RTP header extensions should be automatically enabled, if an implementation is available",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "true",
"mutable": "ready",
"readable": true,
"type": "gboolean",
"writable": true
},
"extensions": {
"blurb": "List of enabled RTP header extensions",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"mutable": "null",
"readable": true,
"type": "GstValueArray",
"writable": false
},
"max-reorder": {
"blurb": "Maximum seqnum reorder before assuming sender has restarted",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "100",
"max": "32767",
"min": "0",
"mutable": "playing",
"readable": true,
"type": "guint",
"writable": true
},
"source-info": {
"blurb": "Add RTP source information as buffer metadata",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "playing",
"readable": true,
"type": "gboolean",
"writable": true
},
"stats": {
"blurb": "Various statistics",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "application/x-rtp-depayload-stats;",
"mutable": "null",
"readable": true,
"type": "GstStructure",
"writable": false
}
},
"signals": {
"add-extension": {
"action": true,
"args": [
{
"name": "arg0",
"type": "GstRTPHeaderExtension"
}
],
"return-type": "void",
"when": "last"
},
"clear-extensions": {
"action": true,
"args": [],
"return-type": "void",
"when": "last"
},
"request-extension": {
"args": [
{
"name": "arg0",
"type": "guint"
},
{
"name": "arg1",
"type": "gchararray"
}
],
"return-type": "GstRTPHeaderExtension",
"when": "last"
}
}
},
"GstRtpBasePay2": {
"hierarchy": [
"GstRtpBasePay2",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"kind": "object",
"properties": {
rtp2: Fix typo on auto-header-extension property name The rtp (de)pay elements use auto-header-extension so the new elements should do the same. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1646>
2024-07-02 08:35:39 +00:00
"auto-header-extension": {
rtp: Add PCMU/PCMA RTP payloader / depayloader elements These come with new generic RTP payloader, RTP raw-ish audio payloader and RTP depayloader base classes. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1424>
2023-10-14 08:23:56 +00:00
"blurb": "Whether RTP header extensions should be automatically enabled, if an implementation is available",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "true",
"mutable": "ready",
"readable": true,
"type": "gboolean",
"writable": true
},
"extensions": {
"blurb": "List of enabled RTP header extensions",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"mutable": "null",
"readable": true,
"type": "GstValueArray",
"writable": false
},
"mtu": {
"blurb": "Maximum size of one RTP packet",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "1400",
"max": "-1",
"min": "28",
"mutable": "playing",
"readable": true,
"type": "guint",
"writable": true
},
"onvif-no-rate-control": {
"blurb": "Enable ONVIF Rate-Control=no timestamping mode",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "ready",
"readable": true,
"type": "gboolean",
"writable": true
},
"pt": {
"blurb": "Payload type of the packets",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
rtp: Port RTP AV1 payloader/depayloader to new base classes Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1472>
2023-11-20 15:02:55 +00:00
"default": "96",
rtp: Add PCMU/PCMA RTP payloader / depayloader elements These come with new generic RTP payloader, RTP raw-ish audio payloader and RTP depayloader base classes. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1424>
2023-10-14 08:23:56 +00:00
"max": "127",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint",
"writable": true
},
"scale-rtptime": {
"blurb": "Whether the RTP timestamp should be scaled with the rate (speed)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "true",
"mutable": "ready",
"readable": true,
"type": "gboolean",
"writable": true
},
"seqnum": {
"blurb": "RTP sequence number of the last packet",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "65535",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": false
},
"seqnum-offset": {
"blurb": "Offset that is added to all RTP sequence numbers (-1 == random)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "-1",
"max": "65535",
"min": "-1",
"mutable": "ready",
"readable": true,
"type": "gint",
"writable": true
},
"source-info": {
"blurb": "Add RTP source information as buffer metadata",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "playing",
"readable": true,
"type": "gboolean",
"writable": true
},
"ssrc": {
"blurb": "SSRC of the packets (-1 == random)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "18446744073709551615",
"max": "4294967295",
"min": "-1",
"mutable": "ready",
"readable": true,
"type": "gint64",
"writable": true
},
"stats": {
"blurb": "Various statistics",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "application/x-rtp-payload-stats;",
"mutable": "null",
"readable": true,
"type": "GstStructure",
"writable": false
},
"timestamp": {
"blurb": "RTP timestamp of the last packet",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "65535",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": false
},
"timestamp-offset": {
"blurb": "Offset that is added to all RTP timestamps (-1 == random)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "18446744073709551615",
"max": "4294967295",
"min": "-1",
"mutable": "ready",
"readable": true,
"type": "gint64",
"writable": true
}
},
"signals": {
"add-extension": {
"action": true,
"args": [
{
"name": "arg0",
"type": "GstRTPHeaderExtension"
}
],
"return-type": "void",
"when": "last"
},
"clear-extensions": {
"action": true,
"args": [],
"return-type": "void",
"when": "last"
},
"request-extension": {
"args": [
{
"name": "arg0",
"type": "guint"
},
{
"name": "arg1",
"type": "gchararray"
}
],
"return-type": "GstRTPHeaderExtension",
"when": "last"
}
}
},
rtpbin2: implement and use synchronization context Co-authored-by: Sebastian Dröge <sebastian@centricular.com> Co-Authored-By: Matthew Waters <matthew@centricular.com> Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
2024-01-04 16:31:36 +00:00
"GstRtpBin2TimestampingMode": {
"kind": "enum",
"values": [
{
"desc": "Use arrival time as timestamp",
"name": "arrival",
"value": "0"
},
{
"desc": "Use RTP timestamps as is",
"name": "rtp",
"value": "1"
},
{
"desc": "Correct skew to synchronize sender and receiver clocks",
"name": "skew",
"value": "2"
}
]
},
rtpgccbwe: Support linear regression based delay estimation In our tests, the slope (found with linear regression) on a history of the (smoothed) accumulated inter-group delays gives a more stable congestion control. In particular, low-end devices becomes less sensitive to spikes in inter-group delay measurements. This flavour of delay based bandwidth estimation with Google Congestion Control is also what Chromium is using. To make it easy to experiment with the new estimator, as well as add support for new ones in the future, also add infrastructure for making delay estimator flavour selectable at runtime. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1566>
2024-05-02 12:13:10 +00:00
"GstRtpGCCBwEEstimator": {
"kind": "enum",
"values": [
{
"desc": "Use Kalman filter",
"name": "kalman",
"value": "0"
},
{
"desc": "Use linear regression slope",
"name": "linear-regression",
"value": "1"
}
]
},
rtp: add mp4gpay Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1551>
2024-04-24 16:33:12 +00:00
"GstRtpMpeg4GenericPayAggregateMode": {
"kind": "enum",
"values": [
{
"desc": "Automatic: zero-latency if upstream is live, otherwise aggregate elementary streams until packet is full.",
"name": "auto",
"value": "-1"
},
{
"desc": "Zero Latency: always send out elementary streams right away, do not wait for more elementary streams to fill a packet.",
"name": "zero-latency",
"value": "0"
},
{
"desc": "Aggregate: collect elementary streams until we have a full packet or the max-ptime limit is hit (if set).",
"name": "aggregate",
"value": "1"
}
]
},
rtp: Add PCMU/PCMA RTP payloader / depayloader elements These come with new generic RTP payloader, RTP raw-ish audio payloader and RTP depayloader base classes. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1424>
2023-10-14 08:23:56 +00:00
"GstRtpPcmauDepay2": {
"hierarchy": [
"GstRtpPcmauDepay2",
"GstRtpBaseDepay2",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"kind": "object"
},
"GstRtpPcmauPay2": {
"hierarchy": [
"GstRtpPcmauPay2",
"GstRtpBaseAudioPay2",
"GstRtpBasePay2",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"kind": "object"
rtp: Add VP8/9 RTP payloader/depayloader Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1487>
2023-12-12 12:53:47 +00:00
},
rtspsrc2: Update rtpbin2 support to use rtprecv and rtpsend USE_RTPBIN2 is now USE_RTP2 because there is no "rtpbin2" now. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
2024-03-14 07:25:34 +00:00
"GstRtpSendProfile": {
"kind": "enum",
"values": [
{
"desc": "AVP profile as specified in RFC 3550",
"name": "avp",
"value": "0"
},
{
"desc": "AVPF profile as specified in RFC 4585",
"name": "avpf",
"value": "1"
}
]
},
rtp: Add VP8/9 RTP payloader/depayloader Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1487>
2023-12-12 12:53:47 +00:00
"GstRtpVp8Pay2FragmentationMode": {
"kind": "enum",
"values": [
{
"desc": "Fit as much into each packet as possible",
"name": "none",
"value": "0"
},
{
"desc": "Make sure that every partition starts at the start of a packet",
"name": "partition-start",
"value": "1"
},
{
"desc": "Create a new packet for every partition",
"name": "every-partition",
"value": "2"
}
]
},
"GstRtpVp8Pay2PictureIdMode": {
"kind": "enum",
"values": [
{
"desc": "No Picture ID",
"name": "none",
"value": "0"
},
{
"desc": "7-bit PictureID",
"name": "7-bit",
"value": "1"
},
{
"desc": "15-bit Picture ID",
"name": "15-bit",
"value": "2"
}
]
},
"GstRtpVp9Pay2PictureIdMode": {
"kind": "enum",
"values": [
{
"desc": "No Picture ID",
"name": "none",
"value": "0"
},
{
"desc": "7-bit PictureID",
"name": "7-bit",
"value": "1"
},
{
"desc": "15-bit Picture ID",
"name": "15-bit",
"value": "2"
}
]
rtp: Add PCMU/PCMA RTP payloader / depayloader elements These come with new generic RTP payloader, RTP raw-ish audio payloader and RTP depayloader base classes. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1424>
2023-10-14 08:23:56 +00:00
}
},
Update docs
2022-10-23 17:35:14 +00:00
"package": "gst-plugin-rtp",
"source": "gst-plugin-rtp",
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2022-08-25 22:30:08 +00:00
"tracers": {},
"url": "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
New RTSP source plugin with live streaming support GST_PLUGIN_FEATURE_RANK=rtspsrc2:1 gst-play-1.0 [URI] Features: * Live streaming N audio and N video - With RTCP-based A/V sync * Lower transports: TCP, UDP, UDP-Multicast * RTP, RTCP SR, RTCP RR * OPTIONS DESCRIBE SETUP PLAY TEARDOWN * Custom UDP socket management, does not use udpsrc/udpsink * Supports both rtpbin and the rtpbin2 rust rewrite - Set USE_RTPBIN2=1 to use rtpbin2 (needs other MRs) * Properties: - protocols selection and priority (NEW!) - location supports rtsp[ut]:// - port-start instead of port-range Co-Authored-by: Tim-Philipp Müller <tim@centricular.com> Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1425>
2024-01-10 10:10:19 +00:00
},
"rsrtsp": {
"description": "GStreamer RTSP Client Plugin",
"elements": {
"rtspsrc2": {
"author": "Nirbheek Chauhan <nirbheek centricular com>",
"description": "Receive audio or video from a network device via the Real Time Streaming Protocol (RTSP) (RFC 2326, 7826)",
"hierarchy": [
"GstRtspSrc2",
"GstBin",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"interfaces": [
"GstChildProxy",
"GstURIHandler"
],
"klass": "Source/Network",
"pad-templates": {
"stream_%%u": {
"caps": "application/x-rtp:\n",
"direction": "src",
"presence": "sometimes"
}
},
"properties": {
"location": {
"blurb": "RTSP server, credentials and media path, e.g. rtsp://user:p4ssw0rd@camera-5.local:8554/h264_1080p30",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
},
"port-start": {
"blurb": "Port number to start allocating client ports for receiving RTP and RTCP data, eg. 3000 (0 = automatic selection)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "-1",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint",
"writable": true
},
"protocols": {
"blurb": "Allowed lower transport protocols, in order of preference",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "udp-mcast,udp,tcp",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
},
rtspsrc2: Allocate a buffer pool for UDP RTP data Control the size with a receive-mtu property Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1425>
2024-01-13 20:50:20 +00:00
"receive-mtu": {
"blurb": "Initial size of buffers to allocate in the buffer pool, will be increased if too small",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "1508",
"max": "-1",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint",
"writable": true
},
New RTSP source plugin with live streaming support GST_PLUGIN_FEATURE_RANK=rtspsrc2:1 gst-play-1.0 [URI] Features: * Live streaming N audio and N video - With RTCP-based A/V sync * Lower transports: TCP, UDP, UDP-Multicast * RTP, RTCP SR, RTCP RR * OPTIONS DESCRIBE SETUP PLAY TEARDOWN * Custom UDP socket management, does not use udpsrc/udpsink * Supports both rtpbin and the rtpbin2 rust rewrite - Set USE_RTPBIN2=1 to use rtpbin2 (needs other MRs) * Properties: - protocols selection and priority (NEW!) - location supports rtsp[ut]:// - port-start instead of port-range Co-Authored-by: Tim-Philipp Müller <tim@centricular.com> Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1425>
2024-01-10 10:10:19 +00:00
"timeout": {
"blurb": "Timeout for network activity, in nanoseconds",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
rtspsrc2: Increase RTP timeout to 5 seconds, matching rtspsrc Also fix some logging. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1425>
2024-01-22 23:34:19 +00:00
"default": "5000000000",
New RTSP source plugin with live streaming support GST_PLUGIN_FEATURE_RANK=rtspsrc2:1 gst-play-1.0 [URI] Features: * Live streaming N audio and N video - With RTCP-based A/V sync * Lower transports: TCP, UDP, UDP-Multicast * RTP, RTCP SR, RTCP RR * OPTIONS DESCRIBE SETUP PLAY TEARDOWN * Custom UDP socket management, does not use udpsrc/udpsink * Supports both rtpbin and the rtpbin2 rust rewrite - Set USE_RTPBIN2=1 to use rtpbin2 (needs other MRs) * Properties: - protocols selection and priority (NEW!) - location supports rtsp[ut]:// - port-start instead of port-range Co-Authored-by: Tim-Philipp Müller <tim@centricular.com> Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1425>
2024-01-10 10:10:19 +00:00
"max": "18446744073709551614",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint64",
"writable": true
}
},
"rank": "none"
}
},
"filename": "gstrsrtsp",
"license": "MPL",
"other-types": {},
"package": "gst-plugin-rtsp",
"source": "gst-plugin-rtsp",
"tracers": {},
"url": "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
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2022-08-25 22:30:08 +00:00
},
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2022-10-23 17:35:14 +00:00
"rstracers": {
"description": "GStreamer Rust tracers plugin",
"elements": {},
"filename": "gstrstracers",
"license": "MPL",
"other-types": {},
"package": "gst-plugin-tracers",
"source": "gst-plugin-tracers",
"tracers": {
"buffer-lateness": {},
tracers: Add a pad push durations tracer This tracer measures the time it takes for a buffer/buffer list push to return. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1506>
2024-03-20 10:23:54 +00:00
"pad-push-timings": {},
tracers: Add a tracer that dumps data flow into .pcap files See documentation for more details Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/879>
2022-09-24 19:47:10 +00:00
"pcap-writer": {},
Update docs
2022-10-23 17:35:14 +00:00
"pipeline-snapshot": {},
"queue-levels": {}
},
"url": "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
},
"rsvideofx": {
"description": "GStreamer Rust Video Effects Plugin",
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"elements": {
Update docs
2022-10-23 17:35:14 +00:00
"colordetect": {
"author": "Philippe Normand <philn@igalia.com>",
"description": "Detects the dominant color of a video",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"hierarchy": [
Update docs
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"GstColorDetect",
"GstBaseTransform",
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"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
Update docs
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"klass": "Filter/Video",
"long-name": "Dominant color detection",
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"pad-templates": {
"sink": {
Update docs
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"caps": "video/x-raw:\n format: { RGB, RGBA, ARGB, BGR, BGRA }\n width: [ 1, 2147483647 ]\n height: [ 1, 2147483647 ]\n framerate: [ 0/1, 2147483647/1 ]\n",
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"direction": "sink",
"presence": "always"
},
"src": {
Update docs
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"caps": "video/x-raw:\n format: { RGB, RGBA, ARGB, BGR, BGRA }\n width: [ 1, 2147483647 ]\n height: [ 1, 2147483647 ]\n framerate: [ 0/1, 2147483647/1 ]\n",
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"direction": "src",
"presence": "always"
}
},
"properties": {
Update docs
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"max-colors": {
"blurb": "Actual colors count can be lower depending on the image",
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"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
Update docs
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"default": "2",
"max": "255",
"min": "2",
"mutable": "playing",
"readable": true,
"type": "guint",
"writable": true
},
"quality": {
"blurb": "A step in pixels to improve performance",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "10",
"max": "10",
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"min": "0",
"mutable": "playing",
"readable": true,
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"type": "guint",
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"writable": true
Update docs
2022-10-23 17:35:14 +00:00
}
},
"rank": "none"
},
"roundedcorners": {
"author": "Sanchayan Maity <sanchayan@asymptotic.io>",
"description": "Adds rounded corners to video",
"hierarchy": [
"GstRoundedCorners",
"GstBaseTransform",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Filter/Effect/Converter/Video",
"long-name": "Rounded Corners",
"pad-templates": {
"sink": {
"caps": "video/x-raw:\n format: I420\n width: [ 1, 2147483647 ]\n height: [ 1, 2147483647 ]\n framerate: [ 0/1, 2147483647/1 ]\n",
"direction": "sink",
"presence": "always"
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},
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"src": {
"caps": "video/x-raw:\n format: { I420, A420 }\n width: [ 1, 2147483647 ]\n height: [ 1, 2147483647 ]\n framerate: [ 0/1, 2147483647/1 ]\n",
"direction": "src",
"presence": "always"
}
},
"properties": {
"border-radius-px": {
"blurb": "Draw rounded corners with given border radius",
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"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
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"default": "0",
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"max": "-1",
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"min": "0",
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"mutable": "playing",
"readable": true,
"type": "guint",
"writable": true
Update docs
2022-10-23 17:35:14 +00:00
}
},
"rank": "none"
},
"videocompare": {
"author": "Rafael Caricio <rafael@caricio.com>",
"description": "Compare similarity of video frames",
"hierarchy": [
"GstVideoCompare",
"GstVideoAggregator",
"GstAggregator",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Filter/Video",
"long-name": "Image comparison",
"pad-templates": {
"sink_%%u": {
"caps": "video/x-raw:\n format: { RGB, RGBA }\n width: [ 1, 2147483647 ]\n height: [ 1, 2147483647 ]\n framerate: [ 0/1, 2147483647/1 ]\n",
"direction": "sink",
"presence": "request",
"type": "GstVideoAggregatorPad"
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},
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2022-10-23 17:35:14 +00:00
"src": {
"caps": "video/x-raw:\n format: { RGB, RGBA }\n width: [ 1, 2147483647 ]\n height: [ 1, 2147483647 ]\n framerate: [ 0/1, 2147483647/1 ]\n",
"direction": "src",
"presence": "always",
"type": "GstVideoAggregatorPad"
}
},
"properties": {
"hash-algo": {
"blurb": "Which hashing algorithm to use for image comparisons",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
Update docs
2022-10-23 17:35:14 +00:00
"default": "blockhash (4)",
"mutable": "ready",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"readable": true,
Update docs
2022-10-23 17:35:14 +00:00
"type": "GstVideoCompareHashAlgorithm",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"writable": true
},
Update docs
2022-10-23 17:35:14 +00:00
"max-dist-threshold": {
"blurb": "Maximum distance threshold to emit messages when an image is detected, by default emits only on exact match",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
Update docs
2022-10-23 17:35:14 +00:00
"max": "1.79769e+308",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"min": "0",
Update docs
2022-10-23 17:35:14 +00:00
"mutable": "ready",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"readable": true,
Update docs
2022-10-23 17:35:14 +00:00
"type": "gdouble",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"writable": true
}
Update docs
2022-10-23 17:35:14 +00:00
},
"rank": "none"
}
},
"filename": "gstrsvideofx",
"license": "MPL",
"other-types": {
"GstVideoCompareHashAlgorithm": {
"kind": "enum",
"values": [
{
"desc": "Mean: The Mean hashing algorithm.",
"name": "mean",
"value": "0"
},
{
"desc": "Gradient: The Gradient hashing algorithm.",
"name": "gradient",
"value": "1"
},
{
"desc": "VertGradient: The Vertical-Gradient hashing algorithm.",
"name": "vertgradient",
"value": "2"
},
{
"desc": "DoubleGradient: The Double-Gradient hashing algorithm.",
"name": "doublegradient",
"value": "3"
},
{
"desc": "Blockhash: The Blockhash (block median value perceptual hash) algorithm.",
"name": "blockhash",
"value": "4"
}
]
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
}
},
Update docs
2022-10-23 17:35:14 +00:00
"package": "gst-plugin-videofx",
"source": "gst-plugin-videofx",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"tracers": {},
"url": "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
},
"rswebp": {
Update docs
2022-10-23 17:35:14 +00:00
"description": "GStreamer WebP Plugin",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"elements": {
Update docs
2022-10-23 17:35:14 +00:00
"rswebpdec": {
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"author": "Mathieu Duponchelle <mathieu@centricular.com>",
"description": "Decodes potentially animated WebP images",
"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstRsWebPDec",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Codec/Decoder/Video",
"long-name": "WebP decoder",
"pad-templates": {
"sink": {
"caps": "image/webp:\n",
"direction": "sink",
"presence": "always"
},
"src": {
plugins, examples, tutorials: Use AudioCapsBuilder and VideoCapsBuilder Simplify caps creation code
2022-10-13 18:02:04 +00:00
"caps": "video/x-raw:\n format: RGBA\n width: [ 1, 2147483647 ]\n height: [ 1, 2147483647 ]\n framerate: [ 0/1, 2147483647/1 ]\n",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"direction": "src",
"presence": "always"
}
},
"rank": "primary"
}
},
"filename": "gstrswebp",
"license": "MPL",
"other-types": {},
"package": "gst-plugin-webp",
"source": "gst-plugin-webp",
"tracers": {},
"url": "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
Plug webrtc in
2022-08-16 14:44:41 +00:00
},
"rswebrtc": {
Update docs
2022-10-23 17:35:14 +00:00
"description": "GStreamer plugin for high level WebRTC elements and a simple signaling server",
Plug webrtc in
2022-08-16 14:44:41 +00:00
"elements": {
net/webrtc: implement AWS KVS signaller And expose a wrapper webrtcsink variant, aws-kvs-webrtcsink. This adds support in webrtcsink for processing a consumer offer, instead of producing one. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1114>
2023-03-01 23:01:43 +00:00
"awskvswebrtcsink": {
"author": "Mathieu Duponchelle <mathieu@centricular.com>",
webrtc: extract a BaseWebRTCSink For documentation purposes, AwsKVSWebRTCSink should not inherit from another element. + Mark base class as plugin API and update plugin cache Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1178>
2023-04-13 15:02:18 +00:00
"description": "WebRTC sink with kinesis video streams signaller",
net/webrtc: implement AWS KVS signaller And expose a wrapper webrtcsink variant, aws-kvs-webrtcsink. This adds support in webrtcsink for processing a consumer offer, instead of producing one. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1114>
2023-03-01 23:01:43 +00:00
"hierarchy": [
"GstAwsKvsWebRTCSink",
webrtc: extract a BaseWebRTCSink For documentation purposes, AwsKVSWebRTCSink should not inherit from another element. + Mark base class as plugin API and update plugin cache Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1178>
2023-04-13 15:02:18 +00:00
"GstBaseWebRTCSink",
net/webrtc: implement AWS KVS signaller And expose a wrapper webrtcsink variant, aws-kvs-webrtcsink. This adds support in webrtcsink for processing a consumer offer, instead of producing one. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1114>
2023-03-01 23:01:43 +00:00
"GstBin",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"interfaces": [
"GstChildProxy",
"GstNavigation"
],
"klass": "Sink/Network/WebRTC",
"pad-templates": {
"audio_%%u": {
webrtcsink: encoded streams: address last review comments Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1194>
2023-06-06 13:41:58 +00:00
"caps": "audio/x-raw:\naudio/x-opus:\n",
net/webrtc/janusvr: add JanusVRWebRTCSink plugin/signaller The JanusVRWebRTCSink is a new plugin that integrates with the Video Room plugin of the Janus Gateway, which simplifies WebRTC communication. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1362>
2023-10-16 16:16:52 +00:00
"direction": "sink",
webrtcsink: Added sinkpad with "msid" property This forwards to the webrtcbin sinkpad's msid when specified. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1442>
2024-01-19 21:59:55 +00:00
"presence": "request",
"type": "GstWebRTCSinkPad"
net/webrtc/janusvr: add JanusVRWebRTCSink plugin/signaller The JanusVRWebRTCSink is a new plugin that integrates with the Video Room plugin of the Janus Gateway, which simplifies WebRTC communication. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1362>
2023-10-16 16:16:52 +00:00
},
"video_%%u": {
webrtcsink: Support av1 via nvav1enc, av1enc, and rav1enc Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1572>
2024-05-10 18:40:13 +00:00
"caps": "video/x-raw:\n\nvideo/x-raw(memory:CUDAMemory):\n\nvideo/x-raw(memory:GLMemory):\n\nvideo/x-raw(memory:NVMM):\n\nvideo/x-raw(memory:D3D11Memory):\nvideo/x-vp8:\nvideo/x-h264:\nvideo/x-vp9:\nvideo/x-h265:\nvideo/x-av1:\n",
net/webrtc/janusvr: add JanusVRWebRTCSink plugin/signaller The JanusVRWebRTCSink is a new plugin that integrates with the Video Room plugin of the Janus Gateway, which simplifies WebRTC communication. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1362>
2023-10-16 16:16:52 +00:00
"direction": "sink",
webrtcsink: Added sinkpad with "msid" property This forwards to the webrtcbin sinkpad's msid when specified. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1442>
2024-01-19 21:59:55 +00:00
"presence": "request",
"type": "GstWebRTCSinkPad"
net/webrtc/janusvr: add JanusVRWebRTCSink plugin/signaller The JanusVRWebRTCSink is a new plugin that integrates with the Video Room plugin of the Janus Gateway, which simplifies WebRTC communication. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1362>
2023-10-16 16:16:52 +00:00
}
},
"rank": "none"
},
"janusvrwebrtcsink": {
"author": "Eva Pace <epace@igalia.com>",
"description": "WebRTC sink with Janus Video Room signaller",
"hierarchy": [
"GstJanusVRWebRTCSink",
"GstBaseWebRTCSink",
"GstBin",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"interfaces": [
"GstChildProxy",
"GstNavigation"
],
"klass": "Sink/Network/WebRTC",
"pad-templates": {
"audio_%%u": {
"caps": "audio/x-raw:\naudio/x-opus:\n",
net/webrtc: implement AWS KVS signaller And expose a wrapper webrtcsink variant, aws-kvs-webrtcsink. This adds support in webrtcsink for processing a consumer offer, instead of producing one. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1114>
2023-03-01 23:01:43 +00:00
"direction": "sink",
webrtcsink: Added sinkpad with "msid" property This forwards to the webrtcbin sinkpad's msid when specified. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1442>
2024-01-19 21:59:55 +00:00
"presence": "request",
"type": "GstWebRTCSinkPad"
net/webrtc: implement AWS KVS signaller And expose a wrapper webrtcsink variant, aws-kvs-webrtcsink. This adds support in webrtcsink for processing a consumer offer, instead of producing one. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1114>
2023-03-01 23:01:43 +00:00
},
"video_%%u": {
webrtcsink: Support av1 via nvav1enc, av1enc, and rav1enc Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1572>
2024-05-10 18:40:13 +00:00
"caps": "video/x-raw:\n\nvideo/x-raw(memory:CUDAMemory):\n\nvideo/x-raw(memory:GLMemory):\n\nvideo/x-raw(memory:NVMM):\n\nvideo/x-raw(memory:D3D11Memory):\nvideo/x-vp8:\nvideo/x-h264:\nvideo/x-vp9:\nvideo/x-h265:\nvideo/x-av1:\n",
webrtcsink: Add LiveKit WebRTC sink and signaller Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1262>
2023-06-21 19:54:00 +00:00
"direction": "sink",
webrtcsink: Added sinkpad with "msid" property This forwards to the webrtcbin sinkpad's msid when specified. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1442>
2024-01-19 21:59:55 +00:00
"presence": "request",
"type": "GstWebRTCSinkPad"
webrtcsink: Add LiveKit WebRTC sink and signaller Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1262>
2023-06-21 19:54:00 +00:00
}
},
webrtc: janusvrwebrtcsink: add 'use-string-ids' property Instead of exposing all ids properties as strings, we now have two signaller implementations exposing those properties using their actual type. This API is more natural and save the element and application conversions when using numerical ids (Janus's default). I also removed the 'joined-id' property as it's actually the same id as 'feed-id'. I think it would be better to have a 'janus-state' property or something like that for applications willing to know when the room has been joined. This id is also no longer generated by the element by default, as Janus will take care of generating one if not provided. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1486>
2024-03-05 13:04:43 +00:00
"properties": {
"use-string-ids": {
"blurb": "Use strings instead of u64 for Janus IDs, see strings_ids config option in janus.plugin.videoroom.jcfg",
"conditionally-available": false,
"construct": false,
"construct-only": true,
"controllable": false,
"default": "false",
"mutable": "null",
"readable": true,
"type": "gboolean",
"writable": true
}
},
webrtcsink: Add LiveKit WebRTC sink and signaller Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1262>
2023-06-21 19:54:00 +00:00
"rank": "none"
},
"livekitwebrtcsink": {
"author": "Olivier Crête <olivier.crete@collabora.com>",
"description": "WebRTC sink with LiveKit signaller",
"hierarchy": [
"GstLiveKitWebRTCSink",
"GstBaseWebRTCSink",
"GstBin",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"interfaces": [
"GstChildProxy",
"GstNavigation"
],
"klass": "Sink/Network/WebRTC",
"pad-templates": {
"audio_%%u": {
"caps": "audio/x-raw:\naudio/x-opus:\n",
"direction": "sink",
webrtcsink: Added sinkpad with "msid" property This forwards to the webrtcbin sinkpad's msid when specified. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1442>
2024-01-19 21:59:55 +00:00
"presence": "request",
"type": "GstWebRTCSinkPad"
webrtcsink: Add LiveKit WebRTC sink and signaller Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1262>
2023-06-21 19:54:00 +00:00
},
"video_%%u": {
webrtcsink: Support av1 via nvav1enc, av1enc, and rav1enc Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1572>
2024-05-10 18:40:13 +00:00
"caps": "video/x-raw:\n\nvideo/x-raw(memory:CUDAMemory):\n\nvideo/x-raw(memory:GLMemory):\n\nvideo/x-raw(memory:NVMM):\n\nvideo/x-raw(memory:D3D11Memory):\nvideo/x-vp8:\nvideo/x-h264:\nvideo/x-vp9:\nvideo/x-h265:\nvideo/x-av1:\n",
net/webrtc: implement AWS KVS signaller And expose a wrapper webrtcsink variant, aws-kvs-webrtcsink. This adds support in webrtcsink for processing a consumer offer, instead of producing one. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1114>
2023-03-01 23:01:43 +00:00
"direction": "sink",
webrtcsink: Added sinkpad with "msid" property This forwards to the webrtcbin sinkpad's msid when specified. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1442>
2024-01-19 21:59:55 +00:00
"presence": "request",
"type": "GstWebRTCSinkPad"
net/webrtc: implement AWS KVS signaller And expose a wrapper webrtcsink variant, aws-kvs-webrtcsink. This adds support in webrtcsink for processing a consumer offer, instead of producing one. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1114>
2023-03-01 23:01:43 +00:00
}
},
"rank": "none"
},
webrtcsrc: Added LiveKit source element Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1461>
2024-02-13 18:48:29 +00:00
"livekitwebrtcsrc": {
"author": "Jordan Yelloz <jordan.yelloz@collabora.com>",
"description": "WebRTC source with LiveKit signaller",
"hierarchy": [
"GstLiveKitWebRTCSrc",
"GstBaseWebRTCSrc",
"GstBin",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"interfaces": [
"GstChildProxy"
],
"klass": "Source/Network/WebRTC",
"pad-templates": {
net/webrtc: multi producer support in webrtcsrc - Add a new structure Session - manage each producer using a session - avoid send EOS when a session terminates, instead keep running waiting for any new producer to connect - Maintain a bin element per session - each session bin encapsulates webrtcbin and the decoder if needed as well as the parser and filter if requested by the application (through request-encoded-filter) - this will be helpful to cleanup the session's respective elements when the corresponding producer terminates the session Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1339>
2023-11-10 12:58:53 +00:00
"audio_%%s_%%u": {
"caps": "audio/x-raw(ANY):\napplication/x-rtp:\naudio/x-opus:\n",
"direction": "src",
"presence": "sometimes",
"type": "GstWebRTCSrcPad"
},
webrtcsrc: Added LiveKit source element Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1461>
2024-02-13 18:48:29 +00:00
"audio_%%u": {
"caps": "audio/x-raw(ANY):\napplication/x-rtp:\naudio/x-opus:\n",
"direction": "src",
"presence": "sometimes",
"type": "GstWebRTCSrcPad"
},
net/webrtc: multi producer support in webrtcsrc - Add a new structure Session - manage each producer using a session - avoid send EOS when a session terminates, instead keep running waiting for any new producer to connect - Maintain a bin element per session - each session bin encapsulates webrtcbin and the decoder if needed as well as the parser and filter if requested by the application (through request-encoded-filter) - this will be helpful to cleanup the session's respective elements when the corresponding producer terminates the session Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1339>
2023-11-10 12:58:53 +00:00
"video_%%s_%%u": {
"caps": "video/x-raw(ANY):\napplication/x-rtp:\nvideo/x-vp8:\nvideo/x-h264:\nvideo/x-vp9:\nvideo/x-h265:\nvideo/x-av1:\n",
"direction": "src",
"presence": "sometimes",
"type": "GstWebRTCSrcPad"
},
webrtcsrc: Added LiveKit source element Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1461>
2024-02-13 18:48:29 +00:00
"video_%%u": {
webrtcsink: Support av1 via nvav1enc, av1enc, and rav1enc Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1572>
2024-05-10 18:40:13 +00:00
"caps": "video/x-raw(ANY):\napplication/x-rtp:\nvideo/x-vp8:\nvideo/x-h264:\nvideo/x-vp9:\nvideo/x-h265:\nvideo/x-av1:\n",
webrtcsrc: Added LiveKit source element Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1461>
2024-02-13 18:48:29 +00:00
"direction": "src",
"presence": "sometimes",
"type": "GstWebRTCSrcPad"
}
},
"rank": "none"
},
Plug webrtc in
2022-08-16 14:44:41 +00:00
"webrtcsink": {
"author": "Mathieu Duponchelle <mathieu@centricular.com>",
webrtc: extract a BaseWebRTCSink For documentation purposes, AwsKVSWebRTCSink should not inherit from another element. + Mark base class as plugin API and update plugin cache Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1178>
2023-04-13 15:02:18 +00:00
"description": "WebRTC sink with custom protocol signaller",
Plug webrtc in
2022-08-16 14:44:41 +00:00
"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstWebRTCSink",
webrtc: extract a BaseWebRTCSink For documentation purposes, AwsKVSWebRTCSink should not inherit from another element. + Mark base class as plugin API and update plugin cache Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1178>
2023-04-13 15:02:18 +00:00
"GstBaseWebRTCSink",
Plug webrtc in
2022-08-16 14:44:41 +00:00
"GstBin",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"interfaces": [
"GstChildProxy",
"GstNavigation"
],
"klass": "Sink/Network/WebRTC",
"pad-templates": {
"audio_%%u": {
webrtcsink: encoded streams: address last review comments Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1194>
2023-06-06 13:41:58 +00:00
"caps": "audio/x-raw:\naudio/x-opus:\n",
Plug webrtc in
2022-08-16 14:44:41 +00:00
"direction": "sink",
webrtcsink: Added sinkpad with "msid" property This forwards to the webrtcbin sinkpad's msid when specified. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1442>
2024-01-19 21:59:55 +00:00
"presence": "request",
"type": "GstWebRTCSinkPad"
Plug webrtc in
2022-08-16 14:44:41 +00:00
},
"video_%%u": {
webrtcsink: Support av1 via nvav1enc, av1enc, and rav1enc Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1572>
2024-05-10 18:40:13 +00:00
"caps": "video/x-raw:\n\nvideo/x-raw(memory:CUDAMemory):\n\nvideo/x-raw(memory:GLMemory):\n\nvideo/x-raw(memory:NVMM):\n\nvideo/x-raw(memory:D3D11Memory):\nvideo/x-vp8:\nvideo/x-h264:\nvideo/x-vp9:\nvideo/x-h265:\nvideo/x-av1:\n",
Plug webrtc in
2022-08-16 14:44:41 +00:00
"direction": "sink",
webrtcsink: Added sinkpad with "msid" property This forwards to the webrtcbin sinkpad's msid when specified. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1442>
2024-01-19 21:59:55 +00:00
"presence": "request",
"type": "GstWebRTCSinkPad"
Plug webrtc in
2022-08-16 14:44:41 +00:00
}
},
webrtcsink: expose properties for running web server Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1671>
2024-07-18 19:55:38 +00:00
"properties": {
"run-signalling-server": {
"blurb": "Whether the element should run its own signalling server",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "ready",
"readable": true,
"type": "gboolean",
"writable": true
},
"signalling-server-cert": {
"blurb": "Path to TLS certificate the signalling server should use.\n The certificate should be formatted as PKCS 12",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
"signalling-server-cert-password": {
"blurb": "The password for the certificate the signalling server will use",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
"signalling-server-host": {
"blurb": "Address the signalling server should listen on",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0.0.0.0",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
"signalling-server-port": {
"blurb": "Port the signalling server should listen on",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "8443",
"max": "65535",
"min": "1",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
}
},
webrtc: extract a BaseWebRTCSink For documentation purposes, AwsKVSWebRTCSink should not inherit from another element. + Mark base class as plugin API and update plugin cache Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1178>
2023-04-13 15:02:18 +00:00
"rank": "none"
},
"webrtcsrc": {
"author": "Thibault Saunier <tsaunier@igalia.com>",
"description": "WebRTC src",
"hierarchy": [
"GstWebRTCSrc",
net/webrtc: Extract BaseWebRTCSrc Define a Base for all the webrtcsrc type elements so they can all be derived from it. Similar to base element defined for webrtcsink type elements Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1284>
2023-07-28 13:09:38 +00:00
"GstBaseWebRTCSrc",
webrtc: extract a BaseWebRTCSink For documentation purposes, AwsKVSWebRTCSink should not inherit from another element. + Mark base class as plugin API and update plugin cache Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1178>
2023-04-13 15:02:18 +00:00
"GstBin",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"interfaces": [
"GstChildProxy",
"GstURIHandler"
],
"klass": "Source/Network/WebRTC",
"long-name": "WebRTCSrc",
"pad-templates": {
net/webrtc: multi producer support in webrtcsrc - Add a new structure Session - manage each producer using a session - avoid send EOS when a session terminates, instead keep running waiting for any new producer to connect - Maintain a bin element per session - each session bin encapsulates webrtcbin and the decoder if needed as well as the parser and filter if requested by the application (through request-encoded-filter) - this will be helpful to cleanup the session's respective elements when the corresponding producer terminates the session Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1339>
2023-11-10 12:58:53 +00:00
"audio_%%s_%%u": {
"caps": "audio/x-raw(ANY):\napplication/x-rtp:\naudio/x-opus:\n",
"direction": "src",
"presence": "sometimes",
"type": "GstWebRTCSrcPad"
},
webrtc: extract a BaseWebRTCSink For documentation purposes, AwsKVSWebRTCSink should not inherit from another element. + Mark base class as plugin API and update plugin cache Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1178>
2023-04-13 15:02:18 +00:00
"audio_%%u": {
webrtcsink: encoded streams: address last review comments Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1194>
2023-06-06 13:41:58 +00:00
"caps": "audio/x-raw(ANY):\napplication/x-rtp:\naudio/x-opus:\n",
webrtc: extract a BaseWebRTCSink For documentation purposes, AwsKVSWebRTCSink should not inherit from another element. + Mark base class as plugin API and update plugin cache Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1178>
2023-04-13 15:02:18 +00:00
"direction": "src",
"presence": "sometimes",
"type": "GstWebRTCSrcPad"
},
net/webrtc: multi producer support in webrtcsrc - Add a new structure Session - manage each producer using a session - avoid send EOS when a session terminates, instead keep running waiting for any new producer to connect - Maintain a bin element per session - each session bin encapsulates webrtcbin and the decoder if needed as well as the parser and filter if requested by the application (through request-encoded-filter) - this will be helpful to cleanup the session's respective elements when the corresponding producer terminates the session Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1339>
2023-11-10 12:58:53 +00:00
"video_%%s_%%u": {
"caps": "video/x-raw(ANY):\napplication/x-rtp:\nvideo/x-vp8:\nvideo/x-h264:\nvideo/x-vp9:\nvideo/x-h265:\nvideo/x-av1:\n",
"direction": "src",
"presence": "sometimes",
"type": "GstWebRTCSrcPad"
},
webrtc: extract a BaseWebRTCSink For documentation purposes, AwsKVSWebRTCSink should not inherit from another element. + Mark base class as plugin API and update plugin cache Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1178>
2023-04-13 15:02:18 +00:00
"video_%%u": {
webrtcsink: Support av1 via nvav1enc, av1enc, and rav1enc Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1572>
2024-05-10 18:40:13 +00:00
"caps": "video/x-raw(ANY):\napplication/x-rtp:\nvideo/x-vp8:\nvideo/x-h264:\nvideo/x-vp9:\nvideo/x-h265:\nvideo/x-av1:\n",
webrtc: extract a BaseWebRTCSink For documentation purposes, AwsKVSWebRTCSink should not inherit from another element. + Mark base class as plugin API and update plugin cache Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1178>
2023-04-13 15:02:18 +00:00
"direction": "src",
"presence": "sometimes",
"type": "GstWebRTCSrcPad"
}
},
net/webrtc: Extract BaseWebRTCSrc Define a Base for all the webrtcsrc type elements so they can all be derived from it. Similar to base element defined for webrtcsink type elements Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1284>
2023-07-28 13:09:38 +00:00
"rank": "primary"
webrtcsink: bring in signalling code from whipsink as a signaller Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1168>
2023-04-06 22:41:16 +00:00
},
net/webrtc: rename whipwebrtcsink as whipclientsink add a deprecation warning in whipsink to indicate it should be used only for RTP content add documentation in whipsink code regarding usage and deprecation Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1282>
2023-08-16 06:48:04 +00:00
"whipclientsink": {
webrtcsink: bring in signalling code from whipsink as a signaller Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1168>
2023-04-06 22:41:16 +00:00
"author": "Taruntej Kanakamalla <taruntej@asymptotic.io>",
net/webrtc: rename whipwebrtcsink as whipclientsink add a deprecation warning in whipsink to indicate it should be used only for RTP content add documentation in whipsink code regarding usage and deprecation Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1282>
2023-08-16 06:48:04 +00:00
"description": "WebRTC sink with WHIP client signaller",
webrtcsink: bring in signalling code from whipsink as a signaller Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1168>
2023-04-06 22:41:16 +00:00
"hierarchy": [
"GstWhipWebRTCSink",
"GstBaseWebRTCSink",
"GstBin",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"interfaces": [
"GstChildProxy",
"GstNavigation"
],
"klass": "Sink/Network/WebRTC",
"pad-templates": {
"audio_%%u": {
"caps": "audio/x-raw:\naudio/x-opus:\n",
"direction": "sink",
webrtcsink: Added sinkpad with "msid" property This forwards to the webrtcbin sinkpad's msid when specified. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1442>
2024-01-19 21:59:55 +00:00
"presence": "request",
"type": "GstWebRTCSinkPad"
webrtcsink: bring in signalling code from whipsink as a signaller Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1168>
2023-04-06 22:41:16 +00:00
},
"video_%%u": {
webrtcsink: Support av1 via nvav1enc, av1enc, and rav1enc Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1572>
2024-05-10 18:40:13 +00:00
"caps": "video/x-raw:\n\nvideo/x-raw(memory:CUDAMemory):\n\nvideo/x-raw(memory:GLMemory):\n\nvideo/x-raw(memory:NVMM):\n\nvideo/x-raw(memory:D3D11Memory):\nvideo/x-vp8:\nvideo/x-h264:\nvideo/x-vp9:\nvideo/x-h265:\nvideo/x-av1:\n",
webrtcsink: bring in signalling code from whipsink as a signaller Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1168>
2023-04-06 22:41:16 +00:00
"direction": "sink",
webrtcsink: Added sinkpad with "msid" property This forwards to the webrtcbin sinkpad's msid when specified. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1442>
2024-01-19 21:59:55 +00:00
"presence": "request",
"type": "GstWebRTCSinkPad"
webrtcsink: bring in signalling code from whipsink as a signaller Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1168>
2023-04-06 22:41:16 +00:00
}
},
"rank": "none"
net/webrtc: add whipserversrc Implement new signaller WhipServerSignaller - an http server using 'warp' - handlers for the POST, OPTIONS, PATCH and DELETE - fixed path `/whip/endpoint` as the URI - fixed value 'whip-client' as the producer peer id - fixed resource url `/whip/resource/whip-client` Derive whipserversrc element from BaseWebRTCSrc - implement constructed method for ObjectImpl to set non-default signaller, i.e., WhipServerSignaller - bind the properties stun-server and turn-servers to those on the Signaller Connect to 'webrtcbin-ready' signal in the constructor of WhipServerSignaller - it will be emitted by the webrtcsrc when the webrtcbin element is ready - the closure for this signal will in turn connect to webrtcbin's ice-gathering-state and perform send with the answer sdp via the channel - the WhipServer will hold its HTTP response in POST handler until this signal is received or timeout which happens early Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1284>
2023-07-19 05:05:33 +00:00
},
"whipserversrc": {
"author": "Taruntej Kanakamalla <taruntej@asymptotic.io>",
"description": "WebRTC source element using WHIP Server as the signaller",
"hierarchy": [
"GstWhipServerSrc",
"GstBaseWebRTCSrc",
"GstBin",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"interfaces": [
"GstChildProxy"
],
"klass": "Source/Network/WebRTC",
"pad-templates": {
net/webrtc: multi producer support in webrtcsrc - Add a new structure Session - manage each producer using a session - avoid send EOS when a session terminates, instead keep running waiting for any new producer to connect - Maintain a bin element per session - each session bin encapsulates webrtcbin and the decoder if needed as well as the parser and filter if requested by the application (through request-encoded-filter) - this will be helpful to cleanup the session's respective elements when the corresponding producer terminates the session Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1339>
2023-11-10 12:58:53 +00:00
"audio_%%s_%%u": {
"caps": "audio/x-raw(ANY):\napplication/x-rtp:\naudio/x-opus:\n",
"direction": "src",
"presence": "sometimes",
"type": "GstWebRTCSrcPad"
},
net/webrtc: add whipserversrc Implement new signaller WhipServerSignaller - an http server using 'warp' - handlers for the POST, OPTIONS, PATCH and DELETE - fixed path `/whip/endpoint` as the URI - fixed value 'whip-client' as the producer peer id - fixed resource url `/whip/resource/whip-client` Derive whipserversrc element from BaseWebRTCSrc - implement constructed method for ObjectImpl to set non-default signaller, i.e., WhipServerSignaller - bind the properties stun-server and turn-servers to those on the Signaller Connect to 'webrtcbin-ready' signal in the constructor of WhipServerSignaller - it will be emitted by the webrtcsrc when the webrtcbin element is ready - the closure for this signal will in turn connect to webrtcbin's ice-gathering-state and perform send with the answer sdp via the channel - the WhipServer will hold its HTTP response in POST handler until this signal is received or timeout which happens early Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1284>
2023-07-19 05:05:33 +00:00
"audio_%%u": {
"caps": "audio/x-raw(ANY):\napplication/x-rtp:\naudio/x-opus:\n",
"direction": "src",
"presence": "sometimes",
"type": "GstWebRTCSrcPad"
},
net/webrtc: multi producer support in webrtcsrc - Add a new structure Session - manage each producer using a session - avoid send EOS when a session terminates, instead keep running waiting for any new producer to connect - Maintain a bin element per session - each session bin encapsulates webrtcbin and the decoder if needed as well as the parser and filter if requested by the application (through request-encoded-filter) - this will be helpful to cleanup the session's respective elements when the corresponding producer terminates the session Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1339>
2023-11-10 12:58:53 +00:00
"video_%%s_%%u": {
"caps": "video/x-raw(ANY):\napplication/x-rtp:\nvideo/x-vp8:\nvideo/x-h264:\nvideo/x-vp9:\nvideo/x-h265:\nvideo/x-av1:\n",
"direction": "src",
"presence": "sometimes",
"type": "GstWebRTCSrcPad"
},
net/webrtc: add whipserversrc Implement new signaller WhipServerSignaller - an http server using 'warp' - handlers for the POST, OPTIONS, PATCH and DELETE - fixed path `/whip/endpoint` as the URI - fixed value 'whip-client' as the producer peer id - fixed resource url `/whip/resource/whip-client` Derive whipserversrc element from BaseWebRTCSrc - implement constructed method for ObjectImpl to set non-default signaller, i.e., WhipServerSignaller - bind the properties stun-server and turn-servers to those on the Signaller Connect to 'webrtcbin-ready' signal in the constructor of WhipServerSignaller - it will be emitted by the webrtcsrc when the webrtcbin element is ready - the closure for this signal will in turn connect to webrtcbin's ice-gathering-state and perform send with the answer sdp via the channel - the WhipServer will hold its HTTP response in POST handler until this signal is received or timeout which happens early Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1284>
2023-07-19 05:05:33 +00:00
"video_%%u": {
webrtcsink: Support av1 via nvav1enc, av1enc, and rav1enc Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1572>
2024-05-10 18:40:13 +00:00
"caps": "video/x-raw(ANY):\napplication/x-rtp:\nvideo/x-vp8:\nvideo/x-h264:\nvideo/x-vp9:\nvideo/x-h265:\nvideo/x-av1:\n",
net/webrtc: add whipserversrc Implement new signaller WhipServerSignaller - an http server using 'warp' - handlers for the POST, OPTIONS, PATCH and DELETE - fixed path `/whip/endpoint` as the URI - fixed value 'whip-client' as the producer peer id - fixed resource url `/whip/resource/whip-client` Derive whipserversrc element from BaseWebRTCSrc - implement constructed method for ObjectImpl to set non-default signaller, i.e., WhipServerSignaller - bind the properties stun-server and turn-servers to those on the Signaller Connect to 'webrtcbin-ready' signal in the constructor of WhipServerSignaller - it will be emitted by the webrtcsrc when the webrtcbin element is ready - the closure for this signal will in turn connect to webrtcbin's ice-gathering-state and perform send with the answer sdp via the channel - the WhipServer will hold its HTTP response in POST handler until this signal is received or timeout which happens early Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1284>
2023-07-19 05:05:33 +00:00
"direction": "src",
"presence": "sometimes",
"type": "GstWebRTCSrcPad"
}
},
"rank": "primary"
webrtc: extract a BaseWebRTCSink For documentation purposes, AwsKVSWebRTCSink should not inherit from another element. + Mark base class as plugin API and update plugin cache Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1178>
2023-04-13 15:02:18 +00:00
}
},
"filename": "gstrswebrtc",
"license": "MPL-2.0",
"other-types": {
"GstBaseWebRTCSink": {
"hierarchy": [
"GstBaseWebRTCSink",
"GstBin",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"interfaces": [
"GstChildProxy",
"GstNavigation"
],
"kind": "object",
Plug webrtc in
2022-08-16 14:44:41 +00:00
"properties": {
"audio-caps": {
webrtc: add raw payload support This commit adds support for raw payloads such as L24 audio to `webrtcsink` & `webrtcsrc`. Most changes take place within the `Codec` helper structure: * A `Codec` can now advertise a depayloader. This also ensures that a format not only can be decoded when necessary, but it can also be depayloaded in the first place. * It is possible to declare raw `Codec`s, meaning that their caps are compatible with a payloader and a depayloader without the need for an encoder and decoder. * Previous accessor `has_decoder` was renamed as `can_be_received` to account for codecs which can be handled by an available depayloader with or without the need for a decoder. * New codecs were added for the following formats: * L24, L16, L8 audio. * RAW video. The `webrtc-precise-sync` examples were updated to demonstrate streaming of raw audio or video. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1501>
2024-07-05 15:38:30 +00:00
"blurb": "Governs what audio codecs will be proposed. Valid values: [audio/x-opus; audio/x-raw, format=(string)S24BE, layout=(string)interleaved, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]; audio/x-raw, format=(string)S16BE, layout=(string)interleaved, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]; audio/x-raw, format=(string)U8, layout=(string)interleaved, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]]",
Plug webrtc in
2022-08-16 14:44:41 +00:00
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "audio/x-opus",
"mutable": "ready",
"readable": true,
"type": "GstCaps",
"writable": true
},
"congestion-control": {
"blurb": "Defines how congestion is controlled, if at all",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "gcc (2)",
"mutable": "ready",
"readable": true,
"type": "GstWebRTCSinkCongestionControl",
"writable": true
},
webrtc: add RFC 7273 support This commit implements [RFC 7273] (NTP & PTP clock signalling & synchronization) for `webrtcsink` by adding the "ts-refclk" & "mediaclk" SDP media attributes to identify the clock. These attributes are handled by `rtpjitterbuffer` on the consumer side. They MUST be part of the SDP offer. When used with an NTP or PTP clock, "mediaclk" indicates the RTP offset at the clock's origin. Because the payloaders are not instantiated when the offer is sent to the consumer, the RTP offset is set to 0 and the payloader `timstamp-offset`s are set accordingly when they are created. The `webrtc-precise-sync` examples were updated to be able to start with an NTP (default), a PTP or the system clock (on the receiver only). The rtp jitter buffer will synchronize with the clock signalled in the SDP offer provided the sender is started with `--do-clock-signalling` & the receiver with `--expect-clock-signalling`. [RFC 7273]: https://datatracker.ietf.org/doc/html/rfc7273 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1500>
2024-04-12 12:18:09 +00:00
"do-clock-signalling": {
"blurb": "Whether PTP or NTP clock & RTP/clock offset should be signalled according to RFC 7273",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "ready",
"readable": true,
"type": "gboolean",
"writable": true
},
Plug webrtc in
2022-08-16 14:44:41 +00:00
"do-fec": {
"blurb": "Whether the element should negotiate and send FEC data",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "true",
"mutable": "ready",
"readable": true,
"type": "gboolean",
"writable": true
},
"do-retransmission": {
"blurb": "Whether the element should offer to honor retransmission requests",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "true",
"mutable": "ready",
"readable": true,
"type": "gboolean",
"writable": true
},
gstwebrtc-api example: add support for requesting mix matrix This is one example of how a consumer might send over custom upstream event requests to the producer. As webrtcsink will deserialize numbers in priority as integers, we need a custom stringifying function to ensure members of the matrix array are indeed serialized with the floating point. An optional stringifier parameter is thus added to the sendControlRequest API. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1711>
2024-08-15 14:27:48 +00:00
"enable-control-data-channel": {
"blurb": "Enable receiving arbitrary events through data channel",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "ready",
"readable": true,
"type": "gboolean",
"writable": true
},
Plug webrtc in
2022-08-16 14:44:41 +00:00
"enable-data-channel-navigation": {
"blurb": "Enable navigation events through a dedicated WebRTCDataChannel",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "ready",
"readable": true,
"type": "gboolean",
"writable": true
},
webrtcsink: Add ice-transport-policy option Can be used to force relay ICE candidates, ensuring TURN server is used. Proxy to the corresponding setting in webrtcbin, Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1143>
2023-03-23 10:27:43 +00:00
"ice-transport-policy": {
"blurb": "The policy to apply for ICE transport",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "all (0)",
"mutable": "ready",
"readable": true,
"type": "GstWebRTCICETransportPolicy",
"writable": true
},
Plug webrtc in
2022-08-16 14:44:41 +00:00
"max-bitrate": {
chore(webrtcsink): fix max-bitrate blurb and nick Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1150>
2023-03-28 05:11:05 +00:00
"blurb": "Maximum bitrate to use (in bit/sec) when computing it through the congestion control algorithm",
Plug webrtc in
2022-08-16 14:44:41 +00:00
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "8192000",
"max": "-1",
"min": "1",
"mutable": "ready",
"readable": true,
"type": "guint",
"writable": true
},
"meta": {
"blurb": "Free form metadata about the producer",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"mutable": "null",
"readable": true,
"type": "GstStructure",
"writable": true
},
"min-bitrate": {
"blurb": "Minimal bitrate to use (in bit/sec) when computing it through the congestion control algorithm",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "1000",
"max": "-1",
"min": "1",
"mutable": "ready",
"readable": true,
"type": "guint",
"writable": true
},
webrtcsink: expose properties for running web server Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1671>
2024-07-18 19:55:38 +00:00
"run-web-server": {
"blurb": "Whether the element should run a web server",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "ready",
"readable": true,
"type": "gboolean",
"writable": true
},
webrtcsink: update docs for property and signal changes Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1141>
2023-04-06 23:54:28 +00:00
"signaller": {
"blurb": "The Signallable object to use to handle WebRTC Signalling",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"mutable": "ready",
"readable": true,
"type": "GstRSWebRTCSignallableIface",
"writable": false
},
Plug webrtc in
2022-08-16 14:44:41 +00:00
"start-bitrate": {
"blurb": "Start bitrate to use (in bit/sec)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "2048000",
"max": "-1",
"min": "1",
"mutable": "ready",
"readable": true,
"type": "guint",
"writable": true
},
"stats": {
"blurb": "Statistics for the current consumers",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "application/x-webrtcsink-stats;",
"mutable": "null",
"readable": true,
"type": "GstStructure",
"writable": false
},
"stun-server": {
"blurb": "The STUN server of the form stun://hostname:port",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "stun://stun.l.google.com:19302",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
webrtcsink: Make the turn-server prop a `turn-servers` list So that we can simply specify several turn servers at once Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/973>
2022-11-09 18:44:02 +00:00
"turn-servers": {
"blurb": "The TURN servers of the form <\"turn(s)://username:password@host:port\", \"turn(s)://username1:password1@host1:port1\">",
Plug webrtc in
2022-08-16 14:44:41 +00:00
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
webrtcsink: Make the turn-server prop a `turn-servers` list So that we can simply specify several turn servers at once Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/973>
2022-11-09 18:44:02 +00:00
"mutable": "ready",
Plug webrtc in
2022-08-16 14:44:41 +00:00
"readable": true,
webrtcsink: Make the turn-server prop a `turn-servers` list So that we can simply specify several turn servers at once Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/973>
2022-11-09 18:44:02 +00:00
"type": "GstValueArray",
Plug webrtc in
2022-08-16 14:44:41 +00:00
"writable": true
},
"video-caps": {
webrtc: add raw payload support This commit adds support for raw payloads such as L24 audio to `webrtcsink` & `webrtcsrc`. Most changes take place within the `Codec` helper structure: * A `Codec` can now advertise a depayloader. This also ensures that a format not only can be decoded when necessary, but it can also be depayloaded in the first place. * It is possible to declare raw `Codec`s, meaning that their caps are compatible with a payloader and a depayloader without the need for an encoder and decoder. * Previous accessor `has_decoder` was renamed as `can_be_received` to account for codecs which can be handled by an available depayloader with or without the need for a decoder. * New codecs were added for the following formats: * L24, L16, L8 audio. * RAW video. The `webrtc-precise-sync` examples were updated to demonstrate streaming of raw audio or video. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1501>
2024-07-05 15:38:30 +00:00
"blurb": "Governs what video codecs will be proposed. Valid values: [video/x-vp8; video/x-h264; video/x-vp9; video/x-h265; video/x-av1; video/x-raw, format=(string){ RGB, RGBA, BGR, BGRA, AYUV, UYVY, I420, Y41B, UYVP }, width=(int)[ 1, 32767 ], height=(int)[ 1, 32767 ]]",
Plug webrtc in
2022-08-16 14:44:41 +00:00
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
webrtcsink: Support av1 via nvav1enc, av1enc, and rav1enc Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1572>
2024-05-10 18:40:13 +00:00
"default": "video/x-vp8; video/x-h264; video/x-vp9; video/x-h265; video/x-av1",
Plug webrtc in
2022-08-16 14:44:41 +00:00
"mutable": "ready",
"readable": true,
"type": "GstCaps",
"writable": true
webrtcsink: expose properties for running web server Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1671>
2024-07-18 19:55:38 +00:00
},
"web-server-cert": {
"blurb": "Path to TLS certificate the web server should use.\n The certificate should be formatted as PEM",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
"web-server-directory": {
"blurb": "The directory the web server should serve",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "gstwebrtc-api/dist",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
"web-server-host-addr": {
"blurb": "Address the web server should listen on",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "http://127.0.0.1:8080/",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
"web-server-key": {
"blurb": "Path to private encryption key the web server should use.\n The key should be formatted as PEM",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
"web-server-path": {
"blurb": "The root path for the web server",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
Plug webrtc in
2022-08-16 14:44:41 +00:00
}
},
"signals": {
"consumer-added": {
"args": [
{
"name": "arg0",
"type": "gchararray"
},
{
"name": "arg1",
"type": "GstElement"
}
],
"return-type": "void",
webrtcsink: expose consumer-pipeline-created signal This signal is emitted as soon as the pipeline for each consumer is created, and can be used by applications that require a greater level of control over webrtcsink's internals. An example is also provided to demonstrate usage Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1220>
2023-05-23 22:10:03 +00:00
"when": "last"
},
"consumer-pipeline-created": {
"args": [
{
"name": "arg0",
"type": "gchararray"
},
{
"name": "arg1",
"type": "GstPipeline"
}
],
"return-type": "void",
Plug webrtc in
2022-08-16 14:44:41 +00:00
"when": "last"
},
"consumer-removed": {
"args": [
{
"name": "arg0",
"type": "gchararray"
},
{
"name": "arg1",
"type": "GstElement"
}
],
"return-type": "void",
"when": "last"
},
"encoder-setup": {
"args": [
{
"name": "arg0",
"type": "gchararray"
},
{
"name": "arg1",
"type": "gchararray"
},
{
"name": "arg2",
"type": "GstElement"
}
],
"return-type": "gboolean",
"when": "last"
},
"get-sessions": {
"action": true,
"args": [],
"return-type": "GStrv",
"when": "last"
net/webrtc: src: add signal "request-encoded-filter" Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1202>
2023-05-09 10:02:15 +00:00
},
webrtcsink: add payloader-setup signal Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1389>
2023-11-08 00:23:30 +00:00
"payloader-setup": {
"args": [
{
"name": "arg0",
"type": "gchararray"
},
{
"name": "arg1",
"type": "gchararray"
},
{
"name": "arg2",
"type": "GstElement"
}
],
"return-type": "gboolean",
"when": "last"
},
net/webrtc: src: add signal "request-encoded-filter" Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1202>
2023-05-09 10:02:15 +00:00
"request-encoded-filter": {
"args": [
{
"name": "arg0",
"type": "gchararray"
},
{
"name": "arg1",
"type": "gchararray"
},
{
"name": "arg2",
"type": "GstCaps"
}
],
"return-type": "GstElement",
"when": "last"
Plug webrtc in
2022-08-16 14:44:41 +00:00
}
}
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>
2022-07-12 21:13:38 +00:00
},
net/webrtc: Extract BaseWebRTCSrc Define a Base for all the webrtcsrc type elements so they can all be derived from it. Similar to base element defined for webrtcsink type elements Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1284>
2023-07-28 13:09:38 +00:00
"GstBaseWebRTCSrc": {
"hierarchy": [
"GstBaseWebRTCSrc",
"GstBin",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"interfaces": [
"GstChildProxy"
],
"kind": "object",
"properties": {
"audio-codecs": {
webrtc: add raw payload support This commit adds support for raw payloads such as L24 audio to `webrtcsink` & `webrtcsrc`. Most changes take place within the `Codec` helper structure: * A `Codec` can now advertise a depayloader. This also ensures that a format not only can be decoded when necessary, but it can also be depayloaded in the first place. * It is possible to declare raw `Codec`s, meaning that their caps are compatible with a payloader and a depayloader without the need for an encoder and decoder. * Previous accessor `has_decoder` was renamed as `can_be_received` to account for codecs which can be handled by an available depayloader with or without the need for a decoder. * New codecs were added for the following formats: * L24, L16, L8 audio. * RAW video. The `webrtc-precise-sync` examples were updated to demonstrate streaming of raw audio or video. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1501>
2024-07-05 15:38:30 +00:00
"blurb": "Names of audio codecs to be be used during the SDP negotiation. Valid values: [OPUS, L24, L16, L8]",
net/webrtc: Extract BaseWebRTCSrc Define a Base for all the webrtcsrc type elements so they can all be derived from it. Similar to base element defined for webrtcsink type elements Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1284>
2023-07-28 13:09:38 +00:00
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"mutable": "ready",
"readable": true,
"type": "GstValueArray",
"writable": true
},
webrtcsrc: add do-retransmission property Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1509>
2024-03-20 17:48:42 +00:00
"do-retransmission": {
"blurb": "Send retransmission events upstream when a packet is late",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "true",
"mutable": "ready",
"readable": true,
"type": "gboolean",
"writable": true
},
gstwebrtc-api example: add support for requesting mix matrix This is one example of how a consumer might send over custom upstream event requests to the producer. As webrtcsink will deserialize numbers in priority as integers, we need a custom stringifying function to ensure members of the matrix array are indeed serialized with the floating point. An optional stringifier parameter is thus added to the sendControlRequest API. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1711>
2024-08-15 14:27:48 +00:00
"enable-control-data-channel": {
"blurb": "Enable sending control requests through a dedicated WebRTCDataChannel",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "ready",
"readable": true,
"type": "gboolean",
"writable": true
},
net/webrtc: Extract BaseWebRTCSrc Define a Base for all the webrtcsrc type elements so they can all be derived from it. Similar to base element defined for webrtcsink type elements Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1284>
2023-07-28 13:09:38 +00:00
"enable-data-channel-navigation": {
"blurb": "Enable navigation events through a dedicated WebRTCDataChannel",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "ready",
"readable": true,
"type": "gboolean",
"writable": true
},
"meta": {
"blurb": "Free form metadata about the consumer",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"mutable": "ready",
"readable": true,
"type": "GstStructure",
"writable": true
},
"signaller": {
"blurb": "The Signallable object to use to handle WebRTC Signalling",
"conditionally-available": false,
"construct": false,
"construct-only": true,
"controllable": false,
"mutable": "null",
"readable": true,
"type": "GstRSWebRTCSignallableIface",
"writable": true
},
"stun-server": {
"blurb": "The STUN server of the form stun://host:port",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "stun://stun.l.google.com:19302",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
},
"turn-servers": {
"blurb": "The TURN servers of the form <\"turn(s)://username:password@host:port\", \"turn(s)://username1:password1@host1:port1\">",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"mutable": "ready",
"readable": true,
"type": "GstValueArray",
"writable": true
},
"video-codecs": {
webrtc: add raw payload support This commit adds support for raw payloads such as L24 audio to `webrtcsink` & `webrtcsrc`. Most changes take place within the `Codec` helper structure: * A `Codec` can now advertise a depayloader. This also ensures that a format not only can be decoded when necessary, but it can also be depayloaded in the first place. * It is possible to declare raw `Codec`s, meaning that their caps are compatible with a payloader and a depayloader without the need for an encoder and decoder. * Previous accessor `has_decoder` was renamed as `can_be_received` to account for codecs which can be handled by an available depayloader with or without the need for a decoder. * New codecs were added for the following formats: * L24, L16, L8 audio. * RAW video. The `webrtc-precise-sync` examples were updated to demonstrate streaming of raw audio or video. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1501>
2024-07-05 15:38:30 +00:00
"blurb": "Names of video codecs to be be used during the SDP negotiation. Valid values: [VP8, H264, VP9, H265, AV1, RAW]",
net/webrtc: Extract BaseWebRTCSrc Define a Base for all the webrtcsrc type elements so they can all be derived from it. Similar to base element defined for webrtcsink type elements Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1284>
2023-07-28 13:09:38 +00:00
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"mutable": "ready",
"readable": true,
"type": "GstValueArray",
"writable": true
}
},
"signals": {
"request-encoded-filter": {
"args": [
{
"name": "arg0",
"type": "gchararray"
},
{
"name": "arg1",
"type": "gchararray"
},
{
"name": "arg2",
"type": "GstCaps"
}
],
"return-type": "GstElement",
"when": "last"
}
}
},
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>
2022-07-12 21:13:38 +00:00
"GstRSWebRTCSignallableIface": {
"hierarchy": [
"GstRSWebRTCSignallableIface",
"GInterface"
],
"kind": "interface",
"signals": {
webrtcsink: update docs for property and signal changes Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1141>
2023-04-06 23:54:28 +00:00
"consumer-added": {
"args": [
{
"name": "arg0",
"type": "gchararray"
},
{
"name": "arg1",
"type": "GstElement"
}
],
"return-type": "void",
"when": "last"
},
"consumer-removed": {
"args": [
{
"name": "arg0",
"type": "gchararray"
},
{
"name": "arg1",
"type": "GstElement"
}
],
"return-type": "void",
"when": "last"
},
webrtc: signaller iface: fix session-ended vs end-session confusion Session ending is bidirectional: the signaller can tell the sink that a session was ended, and the sink can tell the signaller to end a session. As such, two signals are needed, before this patch the second case was not working as in essence the sink was telling itself that a session was ended, and obviously failing to even find it when trying to end it again. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1167>
2023-04-07 23:28:31 +00:00
"end-session": {
"args": [
{
"name": "arg0",
"type": "gchararray"
}
],
"return-type": "gboolean",
"when": "last"
},
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>
2022-07-12 21:13:38 +00:00
"error": {
"args": [
{
"name": "arg0",
"type": "gchararray"
}
],
"return-type": "void",
"when": "last"
},
"handle-ice": {
webrtcsink: update docs for property and signal changes Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1141>
2023-04-06 23:54:28 +00:00
"action": true,
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>
2022-07-12 21:13:38 +00:00
"args": [
{
"name": "arg0",
"type": "gchararray"
},
{
"name": "arg1",
"type": "guint"
},
{
"name": "arg2",
"type": "gchararray"
},
{
"name": "arg3",
"type": "gchararray"
}
],
"return-type": "void",
"when": "last"
},
webrtcsink: Add a mechanism for SDP munging Unfortunately, server implementations might have odd SDP-related quirks, so let's allow clients a way to work around these oddities themselves. For now, this means that a client can fix up the H.264 profile-level-id as required by Twitch (whose media pipeline is more permissive than the WHIP implementation). Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/516 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1525>
2024-04-03 15:31:55 +00:00
"munge-session-description": {
"args": [
{
"name": "arg0",
"type": "gchararray"
},
{
"name": "arg1",
"type": "GstWebRTCSessionDescription"
}
],
"return-type": "GstWebRTCSessionDescription",
"when": "last"
},
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>
2022-07-12 21:13:38 +00:00
"producer-added": {
"args": [
{
"name": "arg0",
"type": "gchararray"
},
{
"name": "arg1",
"type": "GstStructure"
net/webrtc/signaller: advertise running producers in Listener mode When starting a webrtcsrc-signaller client in Listener mode, only the producers started after the client connection were advertised. All currently running producers were ignored unlike the gstwebrtc-api behavior. This commit now lists all running producers when the client Listener connects and advertises them through the "producer-added" signal. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1296>
2023-08-08 11:14:03 +00:00
},
{
"name": "arg2",
"type": "gboolean"
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>
2022-07-12 21:13:38 +00:00
}
],
"return-type": "void",
"when": "last"
},
"producer-removed": {
"args": [
{
"name": "arg0",
"type": "gchararray"
},
{
"name": "arg1",
"type": "GstStructure"
}
],
"return-type": "void",
"when": "last"
},
"request-meta": {
"args": [],
"return-type": "GstStructure",
"when": "last"
},
webrtcsink: update docs for property and signal changes Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1141>
2023-04-06 23:54:28 +00:00
"send-ice": {
"args": [
{
"name": "arg0",
"type": "gchararray"
},
{
"name": "arg1",
"type": "gchararray"
},
{
"name": "arg2",
"type": "guint"
},
{
"name": "arg3",
"type": "gchararray"
}
],
"return-type": "gboolean",
"when": "last"
},
"send-session-description": {
"args": [
{
"name": "arg0",
"type": "gchararray"
},
{
"name": "arg1",
"type": "GstWebRTCSessionDescription"
}
],
"return-type": "gboolean",
"when": "last"
},
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>
2022-07-12 21:13:38 +00:00
"session-description": {
webrtcsink: update docs for property and signal changes Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1141>
2023-04-06 23:54:28 +00:00
"action": true,
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>
2022-07-12 21:13:38 +00:00
"args": [
{
"name": "arg0",
"type": "gchararray"
},
{
"name": "arg1",
"type": "GstWebRTCSessionDescription"
}
],
"return-type": "void",
"when": "last"
},
"session-ended": {
webrtcsink: update docs for property and signal changes Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1141>
2023-04-06 23:54:28 +00:00
"action": true,
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>
2022-07-12 21:13:38 +00:00
"args": [
{
"name": "arg0",
"type": "gchararray"
}
],
webrtcsink: update docs for property and signal changes Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1141>
2023-04-06 23:54:28 +00:00
"return-type": "gboolean",
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>
2022-07-12 21:13:38 +00:00
"when": "last"
},
"session-requested": {
"args": [
{
"name": "arg0",
"type": "gchararray"
},
{
"name": "arg1",
"type": "gchararray"
webrtcsink: update docs for property and signal changes Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1141>
2023-04-06 23:54:28 +00:00
},
{
"name": "arg2",
"type": "GstWebRTCSessionDescription"
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>
2022-07-12 21:13:38 +00:00
}
],
"return-type": "void",
"when": "last"
},
"session-started": {
"args": [
{
"name": "arg0",
"type": "gchararray"
},
{
"name": "arg1",
"type": "gchararray"
}
],
"return-type": "void",
"when": "last"
},
webrtcsink: update docs for property and signal changes Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1141>
2023-04-06 23:54:28 +00:00
"shutdown": {
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>
2022-07-12 21:13:38 +00:00
"action": true,
"args": [],
"return-type": "void",
"when": "last"
},
webrtcsink: update docs for property and signal changes Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1141>
2023-04-06 23:54:28 +00:00
"start": {
"action": true,
"args": [],
"return-type": "gboolean",
"when": "last"
},
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>
2022-07-12 21:13:38 +00:00
"stop": {
"action": true,
"args": [],
webrtcsink: update docs for property and signal changes Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1141>
2023-04-06 23:54:28 +00:00
"return-type": "gboolean",
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>
2022-07-12 21:13:38 +00:00
"when": "last"
net/webrtc: deprecate consumer-added on the signaller add a new signal webrtcbin-ready in this place doing same thing but can be used for both consumers and producers Please note this change is only to the consumer-added signal on the signaller interface. The consumer-added signal on the webrtcsink is unchanged Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1284>
2023-08-14 06:41:57 +00:00
},
"webrtcbin-ready": {
"args": [
{
"name": "arg0",
"type": "gchararray"
},
{
"name": "arg1",
"type": "GstElement"
}
],
"return-type": "void",
"when": "last"
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>
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}
}
},
Plug webrtc in
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"GstWebRTCSinkCongestionControl": {
"kind": "enum",
"values": [
{
"desc": "Disabled: no congestion control is applied",
"name": "disabled",
"value": "0"
},
{
"desc": "Homegrown: simple sender-side heuristic",
"name": "homegrown",
"value": "1"
},
{
"desc": "Google Congestion Control algorithm",
"name": "gcc",
"value": "2"
}
]
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>
2022-07-12 21:13:38 +00:00
},
webrtcsink: Added sinkpad with "msid" property This forwards to the webrtcbin sinkpad's msid when specified. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1442>
2024-01-19 21:59:55 +00:00
"GstWebRTCSinkPad": {
"hierarchy": [
"GstWebRTCSinkPad",
"GstGhostPad",
"GstProxyPad",
"GstPad",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"kind": "object",
"properties": {
"msid": {
"blurb": "Remote MediaStream ID in use for this pad",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
}
}
},
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>
2022-07-12 21:13:38 +00:00
"GstWebRTCSrcPad": {
"hierarchy": [
"GstWebRTCSrcPad",
"GstGhostPad",
"GstProxyPad",
"GstPad",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"kind": "object"
Plug webrtc in
2022-08-16 14:44:41 +00:00
}
},
"package": "gst-plugin-webrtc",
"source": "gst-plugin-webrtc",
"tracers": {},
"url": "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
},
"sodium": {
Update docs
2022-10-23 17:35:14 +00:00
"description": "GStreamer plugin for libsodium-based file encryption and decryption",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"elements": {
"sodiumdecrypter": {
"author": "Jordan Petridis <jordan@centricular.com>",
"description": "libsodium-based file decrypter",
"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstSodiumDecryptor",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Generic",
"long-name": "Decrypter",
"pad-templates": {
"sink": {
"caps": "application/x-sodium-encrypted:\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "ANY",
"direction": "src",
"presence": "always"
}
},
"properties": {
"receiver-key": {
"blurb": "The private key of the Receiver",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"mutable": "null",
"readable": true,
"type": "GBytes",
"writable": true
},
"sender-key": {
"blurb": "The public key of the Sender",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"mutable": "null",
"readable": false,
"type": "GBytes",
"writable": true
}
},
"rank": "none"
},
"sodiumencrypter": {
"author": "Jordan Petridis <jordan@centricular.com>",
"description": "libsodium-based file encrypter",
"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstSodiumEncrypter",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Generic",
"long-name": "Encrypter",
"pad-templates": {
"sink": {
"caps": "ANY",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "application/x-sodium-encrypted:\n",
"direction": "src",
"presence": "always"
}
},
"properties": {
"block-size": {
"blurb": "The block-size of the chunks",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "32768",
"max": "-1",
"min": "1024",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
"receiver-key": {
"blurb": "The public key of the Receiver",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"mutable": "null",
"readable": true,
"type": "GBytes",
"writable": true
},
"sender-key": {
"blurb": "The private key of the Sender",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"mutable": "null",
"readable": false,
"type": "GBytes",
"writable": true
}
},
"rank": "none"
}
},
"filename": "gstsodium",
"license": "MIT/X11",
"other-types": {},
"package": "gst-plugin-sodium",
"source": "gst-plugin-sodium",
"tracers": {},
"url": "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
},
audio: add speechmatics transcriber Element implemented around the Speechmatics API: <https://docs.speechmatics.com/rt-api-ref> The element also comes with translation support, and offers a similar interface to the one exposed by `awstranscriber`. The Speechmatics service has good accuracy, and can be deployed on premises, offering an advantage over AWS transcribe. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1665>
2024-08-16 11:40:22 +00:00
"speechmatics": {
"description": "GStreamer Speechmatics plugin",
"elements": {
"speechmaticstranscriber": {
"author": "Mathieu Duponchelle <mathieu@centricular.com>",
"description": "Speech to Text filter, using Speechmatics transcribe",
"hierarchy": [
"GstSpeechmaticsTranscriber",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"interfaces": [
"GstChildProxy"
],
"klass": "Audio/Text/Filter",
"pad-templates": {
"sink": {
"caps": "audio/x-raw:\n rate: [ 8000, 48000 ]\n channels: 1\n layout: { (string)interleaved, (string)non-interleaved }\n format: S16LE\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "text/x-raw:\n format: utf8\n",
"direction": "src",
"presence": "always",
"type": "GstSpeechmaticsTranscriberSrcPad"
},
"translate_src_%%u": {
"caps": "text/x-raw:\n format: utf8\n",
"direction": "src",
"presence": "request",
"type": "GstSpeechmaticsTranscriberSrcPad"
}
},
"properties": {
"additional-vocabulary": {
"blurb": "Additional vocabulary speechmatics should use",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"mutable": "ready",
"readable": true,
"type": "GstValueArray",
"writable": true
},
"api-key": {
"blurb": "Speechmatics API Key",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
},
"language-code": {
"blurb": "The Language of the Stream, ISO code",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "en",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
"latency": {
"blurb": "Amount of milliseconds to allow for transcription",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "8000",
"max": "-1",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
"lateness": {
"blurb": "Amount of milliseconds to introduce as lateness",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "-1",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
"url": {
"blurb": "URL of the transcription server",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "ws://0.0.0.0:9000",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
}
},
"rank": "none"
}
},
"filename": "gstspeechmatics",
"license": "Proprietary",
"other-types": {
"GstSpeechmaticsTranscriberSrcPad": {
"hierarchy": [
"GstSpeechmaticsTranscriberSrcPad",
"GstPad",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"kind": "object",
"properties": {
"language-code": {
"blurb": "The Language the Stream must be translated to",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
}
}
}
},
"package": "gst-plugin-speechmatics",
"source": "gst-plugin-speechmatics",
"tracers": {},
"url": "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
},
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"spotify": {
Update docs
2022-10-23 17:35:14 +00:00
"description": "GStreamer Spotify Plugin",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"elements": {
"spotifyaudiosrc": {
"author": "Guillaume Desmottes <guillaume@desmottes.be>",
"description": "Spotify source",
"hierarchy": [
"GstSpotifyAudioSrc",
spotifyaudiosrc: convert to PushSrc Fixes #252
2022-09-10 21:33:48 +00:00
"GstPushSrc",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"GstBaseSrc",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
spotifyaudiosrc: implement URI handler Fix #204
2022-10-17 15:09:16 +00:00
"interfaces": [
"GstURIHandler"
],
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"klass": "Source/Audio",
"long-name": "Spotify source",
"pad-templates": {
"src": {
"caps": "application/ogg:\n",
"direction": "src",
"presence": "always"
}
},
"properties": {
spotifyaudiosrc: Support configurable bitrate Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1073>
2023-02-02 11:05:59 +00:00
"bitrate": {
"blurb": "Spotify audio bitrate in kbit/s",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "160 (1)",
"mutable": "ready",
"readable": true,
"type": "GstRsSpotifyBitrate",
"writable": true
},
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"cache-credentials": {
"blurb": "Directory where to cache Spotify credentials",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
},
"cache-files": {
"blurb": "Directory where to cache downloaded files from Spotify",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
},
"cache-max-size": {
"blurb": "The max allowed size of the cache, in bytes, or 0 to disable the cache limit",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "18446744073709551615",
"min": "0",
"mutable": "ready",
"readable": true,
"type": "guint64",
"writable": true
},
"password": {
spotify: document how to use with non Facebook accounts See discussion on #203. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1490>
2024-03-11 08:09:18 +00:00
"blurb": "Spotify password, Facebook accounts need a device password from https://www.spotify.com/us/account/set-device-password/",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
},
"track": {
"blurb": "Spotify track URI, in the form 'spotify:track:$SPOTIFY_ID'",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
},
"username": {
spotify: document how to use with non Facebook accounts See discussion on #203. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1490>
2024-03-11 08:09:18 +00:00
"blurb": "Spotify username, Facebook accounts need a device username from https://www.spotify.com/us/account/set-device-password/",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
}
},
"rank": "primary"
}
},
"filename": "gstspotify",
"license": "MPL",
spotifyaudiosrc: Support configurable bitrate Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1073>
2023-02-02 11:05:59 +00:00
"other-types": {
"GstRsSpotifyBitrate": {
"kind": "enum",
"values": [
{
"desc": "96 kbit/s",
"name": "96",
"value": "0"
},
{
"desc": "160 kbit/s",
"name": "160",
"value": "1"
},
{
"desc": "320 kbit/s",
"name": "320",
"value": "2"
}
]
}
},
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"package": "gst-plugin-spotify",
"source": "gst-plugin-spotify",
"tracers": {},
"url": "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
},
"textahead": {
Update docs
2022-10-23 17:35:14 +00:00
"description": "GStreamer Plugin for displaying upcoming text buffers ahead of time",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"elements": {
"textahead": {
"author": "Guillaume Desmottes <guillaume@desmottes.be>",
"description": "Display upcoming text buffers ahead",
"hierarchy": [
Update docs
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"GstTextAhead",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Text/Filter",
"long-name": "Text Ahead",
"pad-templates": {
"sink": {
"caps": "text/x-raw:\n format: { utf8, pango-markup }\n",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "text/x-raw:\n format: pango-markup\n",
"direction": "src",
"presence": "always"
}
},
"properties": {
"ahead-attributes": {
"blurb": "Pango span attributes to set on the ahead text",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "size=\"smaller\"",
"mutable": "playing",
"readable": true,
"type": "gchararray",
"writable": true
},
"buffer-start-segment": {
"blurb": "Generate a buffer at the start of the segment with ahead text",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "playing",
"readable": true,
"type": "gboolean",
"writable": true
},
"current-attributes": {
"blurb": "Pango span attributes to set on the text from the current buffer",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "size=\"larger\"",
"mutable": "playing",
"readable": true,
"type": "gchararray",
"writable": true
},
"n-ahead": {
"blurb": "The number of ahead text buffers to display along with the current one",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "1",
"max": "-1",
"min": "0",
"mutable": "playing",
"readable": true,
"type": "guint",
"writable": true
},
textahead: add settings to display previous buffers I'll use this in Karapulse to keep displaying the few previous lyrics rather than having them disappear right away. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1006>
2022-12-11 16:24:40 +00:00
"n-previous": {
"blurb": "The number of previous text buffers to display before the current one",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "-1",
"min": "0",
"mutable": "playing",
"readable": true,
"type": "guint",
"writable": true
},
"previous-attributes": {
"blurb": "Pango span attributes to set on the previous text",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "size=\"smaller\"",
"mutable": "playing",
"readable": true,
"type": "gchararray",
"writable": true
},
Update docs
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"separator": {
"blurb": "Text inserted between each text buffers",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "\n",
"mutable": "playing",
"readable": true,
"type": "gchararray",
"writable": true
}
},
"rank": "primary"
}
},
"filename": "gsttextahead",
"license": "MPL",
"other-types": {},
"package": "gst-plugin-textahead",
"source": "gst-plugin-textahead",
"tracers": {},
"url": "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
},
"textwrap": {
"description": "GStreamer Text Wrap Plugin",
"elements": {
"textwrap": {
"author": "Mathieu Duponchelle <mathieu@centricular.com>",
"description": "Breaks text into fixed-size lines, with optional hyphenation",
"hierarchy": [
"GstTextWrap",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
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"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Text/Filter",
Update docs
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"long-name": "Text Wrapper",
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"pad-templates": {
"sink": {
Update docs
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"caps": "text/x-raw:\n format: utf8\n",
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"direction": "sink",
"presence": "always"
},
"src": {
Update docs
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"caps": "text/x-raw:\n format: utf8\n",
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"direction": "src",
"presence": "always"
}
},
"properties": {
Update docs
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"accumulate-time": {
"blurb": "Cut-off time for input text accumulation (0=do not accumulate)",
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"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
Update docs
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"default": "0",
"max": "18446744073709551614",
"min": "0",
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"mutable": "playing",
"readable": true,
Update docs
2022-10-23 17:35:14 +00:00
"type": "guint64",
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"writable": true
},
Update docs
2022-10-23 17:35:14 +00:00
"columns": {
"blurb": "Maximum number of columns for any given line",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
Update docs
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"default": "32",
"max": "-1",
"min": "1",
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2022-08-25 22:30:08 +00:00
"mutable": "playing",
"readable": true,
Update docs
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"type": "guint",
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2022-08-25 22:30:08 +00:00
"writable": true
},
Update docs
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"dictionary": {
"blurb": "Path to a dictionary to load at runtime to perform hyphenation, see <https://docs.rs/crate/hyphenation/0.7.1> for more information",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
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"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
Update docs
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"default": "NULL",
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"mutable": "playing",
"readable": true,
"type": "gchararray",
"writable": true
},
Update docs
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"lines": {
"blurb": "Split input buffer into output buffers with max lines (0=do not split)",
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"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
Update docs
2022-10-23 17:35:14 +00:00
"default": "0",
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2022-08-25 22:30:08 +00:00
"max": "-1",
"min": "0",
"mutable": "playing",
"readable": true,
"type": "guint",
"writable": true
}
},
Update docs
2022-10-23 17:35:14 +00:00
"rank": "none"
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}
},
Update docs
2022-10-23 17:35:14 +00:00
"filename": "gsttextwrap",
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2022-08-25 22:30:08 +00:00
"license": "MPL",
"other-types": {},
Update docs
2022-10-23 17:35:14 +00:00
"package": "gst-plugin-textwrap",
"source": "gst-plugin-textwrap",
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"tracers": {},
"url": "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
},
"threadshare": {
Update docs
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"description": "GStreamer Threadshare Plugin",
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"elements": {
"ts-appsrc": {
"author": "Sebastian Dröge <sebastian@centricular.com>",
"description": "Thread-sharing app source",
"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstTsAppSrc",
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2022-08-25 22:30:08 +00:00
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Source/Generic",
"long-name": "Thread-sharing app source",
"pad-templates": {
"src": {
"caps": "ANY",
"direction": "src",
"presence": "always"
}
},
"properties": {
"caps": {
"blurb": "Caps to use",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"mutable": "null",
"readable": true,
"type": "GstCaps",
"writable": true
},
"context": {
"blurb": "Context name to share threads with",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
"context-wait": {
"blurb": "Throttle poll loop to run at most once every this many ms",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "1000",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
"do-timestamp": {
"blurb": "Timestamp buffers with the current running time on arrival",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "null",
"readable": true,
"type": "gboolean",
"writable": true
},
"max-buffers": {
"blurb": "Maximum number of buffers to queue up",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "10",
"max": "-1",
"min": "1",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
}
},
"rank": "none",
"signals": {
"end-of-stream": {
"action": true,
"args": [],
"return-type": "gboolean",
"when": "last"
},
"push-buffer": {
"action": true,
"args": [
{
"name": "arg0",
"type": "GstBuffer"
}
],
"return-type": "gboolean",
"when": "last"
}
}
},
ts: introduce ts-audiotestsrc This makes it easy to generate "listenable" signals and to evaluate discontinuities. When the `tuning` feature is activated and the `main-elem` property is set, the element can log the parked duration in %, which is an image of the CPU usage for the ts-context. This commit adds a test mode to `udpsrc-benchmark-sender` which generates default audio buffers from `ts-audiotestsrc`. The `rtp` mode is modified so that it uses `ts-audiotestsrc`.
2022-11-03 11:33:39 +00:00
"ts-audiotestsrc": {
"author": "François Laignel <fengalin@free.fr>",
"description": "Thread-sharing audio test source",
"hierarchy": [
"TsAudioTestSrc",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Source/Test",
"pad-templates": {
"src": {
"caps": "audio/x-raw:\n rate: [ 8000, 2147483646 ]\n channels: [ 1, 2147483646 ]\n layout: interleaved\n format: S16LE\n",
"direction": "src",
"presence": "always"
}
},
"properties": {
"buffer-duration": {
"blurb": "Buffer duration in ms",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "10",
"max": "-1",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
"context": {
"blurb": "Context name to share threads with",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
"context-wait": {
"blurb": "Throttle poll loop to run at most once every this many ms",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "1000",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
"do-timestamp": {
"blurb": "Apply current stream time to buffers",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "null",
"readable": true,
"type": "gboolean",
"writable": true
},
"is-live": {
"blurb": "Whether to act as a live source",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "null",
"readable": true,
"type": "gboolean",
"writable": true
},
"num-buffers": {
"blurb": "Number of buffers to output before sending EOS (-1 = unlimited)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "-1",
"max": "2147483647",
"min": "-1",
"mutable": "null",
"readable": true,
"type": "gint",
"writable": true
}
},
"rank": "none"
},
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2022-08-25 22:30:08 +00:00
"ts-input-selector": {
"author": "Mathieu Duponchelle <mathieu@centricular.com>",
"description": "Simple input selector element",
"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstTsInputSelector",
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2022-08-25 22:30:08 +00:00
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Generic",
"long-name": "Thread-sharing input selector",
"pad-templates": {
"sink_%%u": {
"caps": "ANY",
"direction": "sink",
"presence": "request"
},
"src": {
"caps": "ANY",
"direction": "src",
"presence": "always"
}
},
"properties": {
"active-pad": {
"blurb": "Currently active pad",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"mutable": "playing",
"readable": true,
"type": "GstPad",
"writable": true
},
"context": {
"blurb": "Context name to share threads with",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
"context-wait": {
"blurb": "Throttle poll loop to run at most once every this many ms",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "1000",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
}
},
"rank": "none"
},
"ts-jitterbuffer": {
"author": "Mathieu Duponchelle <mathieu@centricular.com>",
"description": "Simple jitterbuffer",
"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstTsJitterBuffer",
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2022-08-25 22:30:08 +00:00
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Generic",
"long-name": "Thread-sharing jitterbuffer",
"pad-templates": {
"sink": {
"caps": "ANY",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "ANY",
"direction": "src",
"presence": "always"
}
},
"properties": {
"context": {
"blurb": "Context name to share threads with",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
"context-wait": {
"blurb": "Throttle poll loop to run at most once every this many ms",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "1000",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
"do-lost": {
"blurb": "Send an event downstream when a packet is lost",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "null",
"readable": true,
"type": "gboolean",
"writable": true
},
"latency": {
"blurb": "Amount of ms to buffer",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "200",
tsjitterbuffer: Fix latency type when getting property
2022-09-02 14:11:31 +00:00
"max": "-1",
"min": "0",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
"max-dropout-time": {
"blurb": "The maximum time (milliseconds) of missing packets tolerated.",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "60000",
"max": "-1",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
"max-misorder-time": {
"blurb": "The maximum time (milliseconds) of misordered packets tolerated.",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "2000",
"max": "-1",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
"stats": {
"blurb": "Various statistics",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "application/x-rtp-jitterbuffer-stats, num-pushed=(guint64)0, num-lost=(guint64)0, num-late=(guint64)0;",
"mutable": "null",
"readable": true,
"type": "GstStructure",
"writable": false
}
},
"rank": "none",
"signals": {
"clear-pt-map": {
"action": true,
"args": [],
"return-type": "void",
"when": "last"
ts-jitterbuffer: Declare `request-pt-map` signal
2022-09-21 11:31:06 +00:00
},
"request-pt-map": {
"args": [
{
"name": "arg0",
"type": "guint"
}
],
"return-type": "GstCaps",
"when": "last"
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2022-08-25 22:30:08 +00:00
}
}
},
"ts-proxysink": {
"author": "Sebastian Dröge <sebastian@centricular.com>",
"description": "Thread-sharing proxy sink",
"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstTsProxySink",
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2022-08-25 22:30:08 +00:00
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Sink/Generic",
"long-name": "Thread-sharing proxy sink",
"pad-templates": {
"sink": {
"caps": "ANY",
"direction": "sink",
"presence": "always"
}
},
"properties": {
"proxy-context": {
"blurb": "Context name of the proxy to share with",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
}
},
"rank": "none"
},
"ts-proxysrc": {
"author": "Sebastian Dröge <sebastian@centricular.com>",
"description": "Thread-sharing proxy source",
"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstTsProxySrc",
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2022-08-25 22:30:08 +00:00
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Source/Generic",
"long-name": "Thread-sharing proxy source",
"pad-templates": {
"src": {
"caps": "ANY",
"direction": "src",
"presence": "always"
}
},
"properties": {
"context": {
"blurb": "Context name to share threads with",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
"context-wait": {
"blurb": "Throttle poll loop to run at most once every this many ms",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "1000",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
"max-size-buffers": {
"blurb": "Maximum number of buffers to queue (0=unlimited)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "200",
"max": "-1",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
"max-size-bytes": {
"blurb": "Maximum number of bytes to queue (0=unlimited)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "1048576",
"max": "-1",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
"max-size-time": {
"blurb": "Maximum number of nanoseconds to queue (0=unlimited)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "1000000000",
"max": "18446744073709551614",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint64",
"writable": true
},
"proxy-context": {
"blurb": "Context name of the proxy to share with",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
}
},
"rank": "none"
},
"ts-queue": {
"author": "Sebastian Dröge <sebastian@centricular.com>",
"description": "Simple data queue",
"hierarchy": [
Update docs
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"GstTsQueue",
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2022-08-25 22:30:08 +00:00
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Generic",
"long-name": "Thread-sharing queue",
"pad-templates": {
"sink": {
"caps": "ANY",
"direction": "sink",
"presence": "always"
},
"src": {
"caps": "ANY",
"direction": "src",
"presence": "always"
}
},
"properties": {
"context": {
"blurb": "Context name to share threads with",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
"context-wait": {
"blurb": "Throttle poll loop to run at most once every this many ms",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "1000",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
"max-size-buffers": {
"blurb": "Maximum number of buffers to queue (0=unlimited)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "200",
"max": "-1",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
"max-size-bytes": {
"blurb": "Maximum number of bytes to queue (0=unlimited)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "1048576",
"max": "-1",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
"max-size-time": {
"blurb": "Maximum number of nanoseconds to queue (0=unlimited)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "1000000000",
"max": "18446744073709551614",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint64",
"writable": true
}
},
"rank": "none"
},
"ts-tcpclientsrc": {
"author": "Sebastian Dröge <sebastian@centricular.com>, LEE Dongjun <redongjun@gmail.com>",
"description": "Receives data over the network via TCP",
"hierarchy": [
Update docs
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"GstTsTcpClientSrc",
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"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Source/Network",
"long-name": "Thread-sharing TCP client source",
"pad-templates": {
"src": {
"caps": "ANY",
"direction": "src",
"presence": "always"
}
},
"properties": {
"blocksize": {
"blurb": "Size in bytes to read per buffer (-1 = default)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "4096",
"max": "-1",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
"caps": {
"blurb": "Caps to use",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"mutable": "null",
"readable": true,
"type": "GstCaps",
"writable": true
},
"context": {
"blurb": "Context name to share threads with",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
"context-wait": {
"blurb": "Throttle poll loop to run at most once every this many ms",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "1000",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
"host": {
"blurb": "The host IP address to receive packets from",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "127.0.0.1",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
"port": {
"blurb": "Port to receive packets from",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "4953",
"max": "65535",
"min": "0",
"mutable": "null",
"readable": true,
"type": "gint",
"writable": true
}
},
"rank": "none"
},
"ts-udpsink": {
"author": "Mathieu <mathieu@centricular.com>",
"description": "Thread-sharing UDP sink",
"hierarchy": [
Update docs
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"GstTsUdpSink",
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"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Sink/Network",
"long-name": "Thread-sharing UDP sink",
"pad-templates": {
"sink": {
"caps": "ANY",
"direction": "sink",
"presence": "always"
}
},
"properties": {
"auto-multicast": {
"blurb": "Automatically join/leave the multicast groups, FALSE means user has to do it himself",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "true",
"mutable": "null",
"readable": true,
"type": "gboolean",
"writable": true
},
"bind-address": {
"blurb": "Address to bind the socket to",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0.0.0.0",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
"bind-address-v6": {
"blurb": "Address to bind the V6 socket to",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "::",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
"bind-port": {
"blurb": "Port to bind the socket to",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "65535",
"min": "0",
"mutable": "null",
"readable": true,
"type": "gint",
"writable": true
},
"bind-port-v6": {
"blurb": "Port to bind the V6 socket to",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "65535",
"min": "0",
"mutable": "null",
"readable": true,
"type": "gint",
"writable": true
},
"clients": {
"blurb": "A comma separated list of host:port pairs with destinations",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "127.0.0.1:5004",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
"context": {
"blurb": "Context name to share threads with",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
"context-wait": {
"blurb": "Throttle poll loop to run at most once every this many ms",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "1000",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
"loop": {
"blurb": "Set the multicast loop parameter.",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "true",
"mutable": "null",
"readable": true,
"type": "gboolean",
"writable": true
},
"qos-dscp": {
"blurb": "Quality of Service, differentiated services code point (-1 default)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "-1",
"max": "63",
"min": "-1",
"mutable": "null",
"readable": true,
"type": "gint",
"writable": true
},
"socket": {
"blurb": "Socket to use for UDP transmission. (None == allocate)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"mutable": "null",
"readable": true,
"type": "GSocket",
"writable": true
},
"socket-v6": {
"blurb": "IPV6 Socket to use for UDP transmission. (None == allocate)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"mutable": "null",
"readable": true,
"type": "GSocket",
"writable": true
},
"sync": {
"blurb": "Sync on the clock",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "true",
"mutable": "null",
"readable": true,
"type": "gboolean",
"writable": true
},
"ttl": {
"blurb": "Used for setting the unicast TTL parameter",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "64",
"max": "255",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
"ttl-mc": {
"blurb": "Used for setting the multicast TTL parameter",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "1",
"max": "255",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
"used-socket": {
"blurb": "Socket currently in use for UDP transmission. (None = no socket)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"mutable": "null",
"readable": true,
"type": "GSocket",
"writable": false
},
"used-socket-v6": {
"blurb": "V6 Socket currently in use for UDP transmission. (None = no socket)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"mutable": "null",
"readable": true,
"type": "GSocket",
"writable": false
}
},
"rank": "none",
"signals": {
"add": {
"action": true,
"args": [
{
"name": "arg0",
"type": "gchararray"
},
{
"name": "arg1",
"type": "gint"
}
],
"return-type": "void",
"when": "last"
},
"clear": {
"action": true,
"args": [],
"return-type": "void",
"when": "last"
},
"remove": {
"action": true,
"args": [
{
"name": "arg0",
"type": "gchararray"
},
{
"name": "arg1",
"type": "gint"
}
],
"return-type": "void",
"when": "last"
}
}
},
"ts-udpsrc": {
"author": "Sebastian Dröge <sebastian@centricular.com>",
"description": "Receives data over the network via UDP",
"hierarchy": [
Update docs
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"GstTsUdpSrc",
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"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Source/Network",
"long-name": "Thread-sharing UDP source",
"pad-templates": {
"src": {
"caps": "ANY",
"direction": "src",
"presence": "always"
}
},
"properties": {
"address": {
"blurb": "Address/multicast group to listen on",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0.0.0.0",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
threadshare: udpsrc: add buffer-size property Use buffer-size to set the receive buffer size on the socket Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1636>
2024-01-03 12:05:21 +00:00
"buffer-size": {
"blurb": "Size of the kernel receive buffer in bytes, 0=default",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "-1",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
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"caps": {
"blurb": "Caps to use",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"mutable": "null",
"readable": true,
"type": "GstCaps",
"writable": true
},
"context": {
"blurb": "Context name to share threads with",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
"context-wait": {
"blurb": "Throttle poll loop to run at most once every this many ms",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "1000",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
threadshare: udpsrc: add loop property to set multicast loopback Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1636>
2024-06-25 10:04:45 +00:00
"loop": {
"blurb": "Set the multicast loop parameter",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "true",
"mutable": "null",
"readable": true,
"type": "gboolean",
"writable": true
},
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"mtu": {
"blurb": "Maximum expected packet size. This directly defines the allocation size of the receive buffer pool",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "1492",
"max": "2147483647",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
"port": {
"blurb": "Port to listen on",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "5004",
"max": "65535",
"min": "0",
"mutable": "null",
"readable": true,
"type": "gint",
"writable": true
},
"retrieve-sender-address": {
"blurb": "Whether to retrieve the sender address and add it to buffers as meta. Disabling this might result in minor performance improvements in certain scenarios",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "true",
"mutable": "null",
"readable": true,
"type": "gboolean",
"writable": true
},
"reuse": {
"blurb": "Allow reuse of the port",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "true",
"mutable": "null",
"readable": true,
"type": "gboolean",
"writable": true
},
"socket": {
"blurb": "Socket to use for UDP reception. (None == allocate)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"mutable": "null",
"readable": true,
"type": "GSocket",
"writable": true
},
"used-socket": {
"blurb": "Socket currently in use for UDP reception. (None = no socket)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"mutable": "null",
"readable": true,
"type": "GSocket",
"writable": false
}
},
"rank": "none"
}
},
"filename": "gstthreadshare",
"license": "LGPL",
"other-types": {},
"package": "gst-plugin-threadshare",
"source": "gst-plugin-threadshare",
"tracers": {},
"url": "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
},
"togglerecord": {
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"description": "GStreamer Toggle Record Plugin",
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"elements": {
"togglerecord": {
"author": "Sebastian Dröge <sebastian@centricular.com>",
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>
2023-04-27 16:59:19 +00:00
"description": "Valve that ensures multiple streams start/end at the same time. If the input comes from a live stream, when not recording it will be dropped. If it comes from a non-live stream, when not recording it will be blocked.",
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"hierarchy": [
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"GstToggleRecord",
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"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"klass": "Generic",
"long-name": "Toggle Record",
"pad-templates": {
"sink": {
"caps": "ANY",
"direction": "sink",
"presence": "always"
},
"sink_%%u": {
"caps": "ANY",
"direction": "sink",
"presence": "request"
},
"src": {
"caps": "ANY",
"direction": "src",
"presence": "always"
},
"src_%%u": {
"caps": "ANY",
"direction": "src",
"presence": "sometimes"
}
},
"properties": {
"is-live": {
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>
2023-04-27 16:59:19 +00:00
"blurb": "Live output mode: no \"gap eating\", forward incoming segment for live input, create a gap to fill the paused duration for non-live input",
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"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "ready",
"readable": true,
"type": "gboolean",
"writable": true
},
"record": {
"blurb": "Enable/disable recording",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "playing",
"readable": true,
"type": "gboolean",
"writable": true
},
"recording": {
"blurb": "Whether recording is currently taking place",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "null",
"readable": true,
"type": "gboolean",
"writable": false
}
},
"rank": "none"
}
},
"filename": "gsttogglerecord",
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 09:05:00 +00:00
"license": "MPL",
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"other-types": {},
"package": "gst-plugin-togglerecord",
"source": "gst-plugin-togglerecord",
"tracers": {},
"url": "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
},
"uriplaylistbin": {
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"description": "GStreamer Playlist Playback Plugin",
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"elements": {
"uriplaylistbin": {
"author": "Guillaume Desmottes <guillaume.desmottes@onestream.live>",
"description": "Sequentially play uri streams",
"hierarchy": [
"GstUriPlaylistBin",
"GstBin",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"interfaces": [
"GstChildProxy"
],
"klass": "Generic/Source",
"long-name": "Playlist Source",
"pad-templates": {
"audio_%%u": {
"caps": "ANY",
"direction": "src",
"presence": "sometimes"
},
"text_%%u": {
"caps": "ANY",
"direction": "src",
"presence": "sometimes"
},
"video_%%u": {
"caps": "ANY",
"direction": "src",
"presence": "sometimes"
}
},
"properties": {
"current-iteration": {
"blurb": "The index of the current playlist iteration, or 0 if the iterations property is 0 (unlimited playlist)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "-1",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": false
},
"current-uri-index": {
"blurb": "The index from the uris property of the current URI being played",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "0",
"max": "18446744073709551615",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint64",
"writable": false
},
"iterations": {
"blurb": "Number of time the playlist items should be played each (0 = unlimited)",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "1",
"max": "-1",
"min": "0",
uriplaylistbin: allow to change 'iterations' property while playing Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1492>
2024-02-14 12:06:19 +00:00
"mutable": "playing",
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
"readable": true,
"type": "guint",
"writable": true
},
"uris": {
"blurb": "URIs of the medias to play",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"mutable": "ready",
"readable": true,
"type": "GStrv",
"writable": true
}
},
"rank": "none"
}
},
"filename": "gsturiplaylistbin",
"license": "MPL",
"other-types": {},
"package": "gst-plugin-uriplaylistbin",
"source": "gst-plugin-uriplaylistbin",
"tracers": {},
"url": "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
},
whipsink: A GstBin implementation for WHIP Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1410 Created a new plugin 'webrtchttp' to implement all the WebRTC HTTP protocols under /net/webrtc-http directory. WhipSink wraps around 'webrtcbin' with HTTP capabilites to exchange SDP offer/answer so an ICE/DTLS session can be established between the encoder/media producer (WHIP client) and the broadcasting ingestion endpoint (Media Server). Once the ICE/DTLS session is set up, the media will flow unidirectionally from the WHIP client to the broadcasting ingestion endpoint (Media Server). Spec: https://www.ietf.org/archive/id/draft-ietf-wish-whip-04.html
2022-07-19 06:02:47 +00:00
"webrtchttp": {
Add a WebRTC WHEP source element This implements WHEP specification based on https://datatracker.ietf.org/doc/html/draft-murillo-whep-00 and has been tested with Cloudflare. Server offers are likely to be removed from the WHEP specification in upcoming revisions, to avoid compatibility issues. None of the commercial services implementing WHEP support server initiated offers. So we only support client side initiated offers. Follows session setup and tear down as covered in Figure 1, Section 3 of the specification. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
2022-10-24 06:48:31 +00:00
"description": "GStreamer WebRTC Plugin for WebRTC HTTP protocols (WHIP/WHEP)",
whipsink: A GstBin implementation for WHIP Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1410 Created a new plugin 'webrtchttp' to implement all the WebRTC HTTP protocols under /net/webrtc-http directory. WhipSink wraps around 'webrtcbin' with HTTP capabilites to exchange SDP offer/answer so an ICE/DTLS session can be established between the encoder/media producer (WHIP client) and the broadcasting ingestion endpoint (Media Server). Once the ICE/DTLS session is set up, the media will flow unidirectionally from the WHIP client to the broadcasting ingestion endpoint (Media Server). Spec: https://www.ietf.org/archive/id/draft-ietf-wish-whip-04.html
2022-07-19 06:02:47 +00:00
"elements": {
Add a WebRTC WHEP source element This implements WHEP specification based on https://datatracker.ietf.org/doc/html/draft-murillo-whep-00 and has been tested with Cloudflare. Server offers are likely to be removed from the WHEP specification in upcoming revisions, to avoid compatibility issues. None of the commercial services implementing WHEP support server initiated offers. So we only support client side initiated offers. Follows session setup and tear down as covered in Figure 1, Section 3 of the specification. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
2022-10-24 06:48:31 +00:00
"whepsrc": {
"author": "Sanchayan Maity <sanchayan@asymptotic.io>",
"description": "A bin to stream media using the WebRTC HTTP Egress Protocol (WHEP)",
"hierarchy": [
"GstWhepSrc",
"GstBin",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"interfaces": [
"GstChildProxy"
],
"klass": "Source/Network/WebRTC",
"long-name": "WHEP Source Bin",
"pad-templates": {
"src_%%u": {
"caps": "application/x-rtp:\n",
"direction": "src",
"presence": "sometimes"
}
},
"properties": {
"audio-caps": {
"blurb": "Governs what audio codecs will be proposed",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
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>
2024-08-23 14:04:48 +00:00
"default": "application/x-rtp, media=(string)audio, encoding-name=(string)OPUS, payload=(int)96, clock-rate=(int)48000",
Add a WebRTC WHEP source element This implements WHEP specification based on https://datatracker.ietf.org/doc/html/draft-murillo-whep-00 and has been tested with Cloudflare. Server offers are likely to be removed from the WHEP specification in upcoming revisions, to avoid compatibility issues. None of the commercial services implementing WHEP support server initiated offers. So we only support client side initiated offers. Follows session setup and tear down as covered in Figure 1, Section 3 of the specification. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
2022-10-24 06:48:31 +00:00
"mutable": "null",
"readable": true,
"type": "GstCaps",
"writable": true
},
"auth-token": {
"blurb": "Authentication token to use, will be sent in the HTTP Header as 'Bearer <auth-token>'",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
"ice-transport-policy": {
"blurb": "The policy to apply for ICE transport",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "all (0)",
"mutable": "null",
"readable": true,
"type": "GstRsWebRTCICETransportPolicy",
"writable": true
},
"stun-server": {
"blurb": "The STUN server of the form stun://hostname:port",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
"timeout": {
"blurb": "Value in seconds to timeout WHEP endpoint requests (0 = No timeout).",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "15",
"max": "3600",
"min": "0",
"mutable": "null",
"readable": true,
"type": "guint",
"writable": true
},
"turn-server": {
"blurb": "The TURN server of the form turn(s)://username:password@host:port.",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
"use-link-headers": {
webrtchttp: Fix documentation JSON Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
2022-12-05 10:25:50 +00:00
"blurb": "Use link headers to configure STUN/TURN servers if present in WHEP endpoint response.",
Add a WebRTC WHEP source element This implements WHEP specification based on https://datatracker.ietf.org/doc/html/draft-murillo-whep-00 and has been tested with Cloudflare. Server offers are likely to be removed from the WHEP specification in upcoming revisions, to avoid compatibility issues. None of the commercial services implementing WHEP support server initiated offers. So we only support client side initiated offers. Follows session setup and tear down as covered in Figure 1, Section 3 of the specification. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
2022-10-24 06:48:31 +00:00
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "null",
"readable": true,
"type": "gboolean",
"writable": true
},
"video-caps": {
"blurb": "Governs what video codecs will be proposed",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
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>
2024-08-23 14:04:48 +00:00
"default": "application/x-rtp, media=(string)video, payload=(int)101, encoding-name=(string)VP8, clock-rate=(int)90000; application/x-rtp, media=(string)video, payload=(int)102, encoding-name=(string)VP9, clock-rate=(int)90000; application/x-rtp, media=(string)video, payload=(int)103, encoding-name=(string)H264, clock-rate=(int)90000; application/x-rtp, media=(string)video, payload=(int)104, encoding-name=(string)H265, clock-rate=(int)90000; application/x-rtp, media=(string)video, payload=(int)105, encoding-name=(string)AV1, clock-rate=(int)90000",
Add a WebRTC WHEP source element This implements WHEP specification based on https://datatracker.ietf.org/doc/html/draft-murillo-whep-00 and has been tested with Cloudflare. Server offers are likely to be removed from the WHEP specification in upcoming revisions, to avoid compatibility issues. None of the commercial services implementing WHEP support server initiated offers. So we only support client side initiated offers. Follows session setup and tear down as covered in Figure 1, Section 3 of the specification. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
2022-10-24 06:48:31 +00:00
"mutable": "null",
"readable": true,
"type": "GstCaps",
"writable": true
},
"whep-endpoint": {
webrtchttp: Fix documentation JSON Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
2022-12-05 10:25:50 +00:00
"blurb": "The WHEP server endpoint to POST SDP offer to.",
Add a WebRTC WHEP source element This implements WHEP specification based on https://datatracker.ietf.org/doc/html/draft-murillo-whep-00 and has been tested with Cloudflare. Server offers are likely to be removed from the WHEP specification in upcoming revisions, to avoid compatibility issues. None of the commercial services implementing WHEP support server initiated offers. So we only support client side initiated offers. Follows session setup and tear down as covered in Figure 1, Section 3 of the specification. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
2022-10-24 06:48:31 +00:00
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
}
},
"rank": "marginal"
},
whipsink: A GstBin implementation for WHIP Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1410 Created a new plugin 'webrtchttp' to implement all the WebRTC HTTP protocols under /net/webrtc-http directory. WhipSink wraps around 'webrtcbin' with HTTP capabilites to exchange SDP offer/answer so an ICE/DTLS session can be established between the encoder/media producer (WHIP client) and the broadcasting ingestion endpoint (Media Server). Once the ICE/DTLS session is set up, the media will flow unidirectionally from the WHIP client to the broadcasting ingestion endpoint (Media Server). Spec: https://www.ietf.org/archive/id/draft-ietf-wish-whip-04.html
2022-07-19 06:02:47 +00:00
"whipsink": {
"author": "Taruntej Kanakamalla <taruntej@asymptotic.io>",
net/webrtc: rename whipwebrtcsink as whipclientsink add a deprecation warning in whipsink to indicate it should be used only for RTP content add documentation in whipsink code regarding usage and deprecation Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1282>
2023-08-16 06:48:04 +00:00
"description": "A bin to stream RTP media using the WebRTC HTTP Ingestion Protocol (WHIP)",
whipsink: A GstBin implementation for WHIP Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1410 Created a new plugin 'webrtchttp' to implement all the WebRTC HTTP protocols under /net/webrtc-http directory. WhipSink wraps around 'webrtcbin' with HTTP capabilites to exchange SDP offer/answer so an ICE/DTLS session can be established between the encoder/media producer (WHIP client) and the broadcasting ingestion endpoint (Media Server). Once the ICE/DTLS session is set up, the media will flow unidirectionally from the WHIP client to the broadcasting ingestion endpoint (Media Server). Spec: https://www.ietf.org/archive/id/draft-ietf-wish-whip-04.html
2022-07-19 06:02:47 +00:00
"hierarchy": [
Update docs
2022-10-23 17:35:14 +00:00
"GstWhipSink",
whipsink: A GstBin implementation for WHIP Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1410 Created a new plugin 'webrtchttp' to implement all the WebRTC HTTP protocols under /net/webrtc-http directory. WhipSink wraps around 'webrtcbin' with HTTP capabilites to exchange SDP offer/answer so an ICE/DTLS session can be established between the encoder/media producer (WHIP client) and the broadcasting ingestion endpoint (Media Server). Once the ICE/DTLS session is set up, the media will flow unidirectionally from the WHIP client to the broadcasting ingestion endpoint (Media Server). Spec: https://www.ietf.org/archive/id/draft-ietf-wish-whip-04.html
2022-07-19 06:02:47 +00:00
"GstBin",
"GstElement",
"GstObject",
"GInitiallyUnowned",
"GObject"
],
"interfaces": [
"GstChildProxy"
],
"klass": "Sink/Network/WebRTC",
"long-name": "WHIP Sink Bin",
"pad-templates": {
"sink_%%u": {
"caps": "application/x-rtp:\n",
"direction": "sink",
"presence": "request"
}
},
"properties": {
"auth-token": {
"blurb": "Authentication token to use, will be sent in the HTTP Header as 'Bearer <auth-token>'",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
},
webrtchttp: whipsink: Add candidates when sending the offer WHIP endpoint providers like Cloudflare do not support Trickle ICE and need candidates to be send along with the initial offer. Instead of sending the offer in create-offer promise, send it once the ICE candidates have been gathered. While at it add properties to set STUN and TURN server along with the ICE transport policy as at least when testing the Cloudflare WHIP endpoint seems unreachable without it. This has also been observed with Cloudflare provided demos. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
2022-10-27 16:30:57 +00:00
"ice-transport-policy": {
"blurb": "The policy to apply for ICE transport",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "all (0)",
"mutable": "null",
"readable": true,
"type": "GstRsWebRTCICETransportPolicy",
"writable": true
},
"stun-server": {
"blurb": "The STUN server of the form stun://hostname:port",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
"timeout": {
"blurb": "Value in seconds to timeout WHIP endpoint requests (0 = No timeout).",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "15",
"max": "3600",
"min": "0",
webrtchttp: Fix documentation JSON Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
2022-12-05 10:25:50 +00:00
"mutable": "null",
webrtchttp: whipsink: Add candidates when sending the offer WHIP endpoint providers like Cloudflare do not support Trickle ICE and need candidates to be send along with the initial offer. Instead of sending the offer in create-offer promise, send it once the ICE candidates have been gathered. While at it add properties to set STUN and TURN server along with the ICE transport policy as at least when testing the Cloudflare WHIP endpoint seems unreachable without it. This has also been observed with Cloudflare provided demos. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
2022-10-27 16:30:57 +00:00
"readable": true,
"type": "guint",
"writable": true
},
"turn-server": {
"blurb": "The TURN server of the form turn(s)://username:password@host:port.",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "null",
"readable": true,
"type": "gchararray",
"writable": true
},
whipsink: A GstBin implementation for WHIP Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1410 Created a new plugin 'webrtchttp' to implement all the WebRTC HTTP protocols under /net/webrtc-http directory. WhipSink wraps around 'webrtcbin' with HTTP capabilites to exchange SDP offer/answer so an ICE/DTLS session can be established between the encoder/media producer (WHIP client) and the broadcasting ingestion endpoint (Media Server). Once the ICE/DTLS session is set up, the media will flow unidirectionally from the WHIP client to the broadcasting ingestion endpoint (Media Server). Spec: https://www.ietf.org/archive/id/draft-ietf-wish-whip-04.html
2022-07-19 06:02:47 +00:00
"use-link-headers": {
webrtchttp: Fix documentation JSON Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
2022-12-05 10:25:50 +00:00
"blurb": "Use link headers to configure ice-servers from the WHIP server response to the POST request.\n If set to TRUE and the WHIP server returns valid ice-servers,\n this property overrides the ice-servers values set using the stun-server and turn-server properties.",
whipsink: A GstBin implementation for WHIP Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1410 Created a new plugin 'webrtchttp' to implement all the WebRTC HTTP protocols under /net/webrtc-http directory. WhipSink wraps around 'webrtcbin' with HTTP capabilites to exchange SDP offer/answer so an ICE/DTLS session can be established between the encoder/media producer (WHIP client) and the broadcasting ingestion endpoint (Media Server). Once the ICE/DTLS session is set up, the media will flow unidirectionally from the WHIP client to the broadcasting ingestion endpoint (Media Server). Spec: https://www.ietf.org/archive/id/draft-ietf-wish-whip-04.html
2022-07-19 06:02:47 +00:00
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "false",
"mutable": "ready",
"readable": true,
"type": "gboolean",
"writable": true
},
"whip-endpoint": {
"blurb": "The WHIP server endpoint to POST SDP offer to.\n e.g.: https://example.com/whip/endpoint/room1234",
"conditionally-available": false,
"construct": false,
"construct-only": false,
"controllable": false,
"default": "NULL",
"mutable": "ready",
"readable": true,
"type": "gchararray",
"writable": true
}
},
"rank": "marginal"
}
},
"filename": "gstwebrtchttp",
"license": "MPL",
Add a WebRTC WHEP source element This implements WHEP specification based on https://datatracker.ietf.org/doc/html/draft-murillo-whep-00 and has been tested with Cloudflare. Server offers are likely to be removed from the WHEP specification in upcoming revisions, to avoid compatibility issues. None of the commercial services implementing WHEP support server initiated offers. So we only support client side initiated offers. Follows session setup and tear down as covered in Figure 1, Section 3 of the specification. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
2022-10-24 06:48:31 +00:00
"other-types": {
"GstRsWebRTCICETransportPolicy": {
"kind": "enum",
"values": [
{
"desc": "All: get both STUN and TURN candidate pairs",
"name": "all",
"value": "0"
},
{
"desc": "Relay: get only TURN candidate pairs",
"name": "relay",
"value": "1"
webrtchttp: Fix documentation JSON Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
2022-12-05 10:25:50 +00:00
}
Add a WebRTC WHEP source element This implements WHEP specification based on https://datatracker.ietf.org/doc/html/draft-murillo-whep-00 and has been tested with Cloudflare. Server offers are likely to be removed from the WHEP specification in upcoming revisions, to avoid compatibility issues. None of the commercial services implementing WHEP support server initiated offers. So we only support client side initiated offers. Follows session setup and tear down as covered in Figure 1, Section 3 of the specification. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
2022-10-24 06:48:31 +00:00
]
webrtchttp: Fix documentation JSON Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
2022-12-05 10:25:50 +00:00
}
Add a WebRTC WHEP source element This implements WHEP specification based on https://datatracker.ietf.org/doc/html/draft-murillo-whep-00 and has been tested with Cloudflare. Server offers are likely to be removed from the WHEP specification in upcoming revisions, to avoid compatibility issues. None of the commercial services implementing WHEP support server initiated offers. So we only support client side initiated offers. Follows session setup and tear down as covered in Figure 1, Section 3 of the specification. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
2022-10-24 06:48:31 +00:00
},
whipsink: A GstBin implementation for WHIP Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1410 Created a new plugin 'webrtchttp' to implement all the WebRTC HTTP protocols under /net/webrtc-http directory. WhipSink wraps around 'webrtcbin' with HTTP capabilites to exchange SDP offer/answer so an ICE/DTLS session can be established between the encoder/media producer (WHIP client) and the broadcasting ingestion endpoint (Media Server). Once the ICE/DTLS session is set up, the media will flow unidirectionally from the WHIP client to the broadcasting ingestion endpoint (Media Server). Spec: https://www.ietf.org/archive/id/draft-ietf-wish-whip-04.html
2022-07-19 06:02:47 +00:00
"package": "gst-plugin-webrtchttp",
"source": "gst-plugin-webrtchttp",
"tracers": {},
"url": "https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs"
Generate plugins documentation using hotdoc Which will automatically be integrated in gstreamer documentation
2022-08-25 22:30:08 +00:00
}
rtp: add mp4gpay Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1551>
2024-04-24 16:33:12 +00:00
}
Reference in a new issue Copy permalink