gcore 0.13.0__py3-none-any.whl → 0.14.0__py3-none-any.whl

gcore/types/streaming/stream.py

@@ -11,9 +11,12 @@ __all__ = ["Stream"]
 
 class Stream(BaseModel):
     name: str
-    """
-    Stream name. Often used as a human-readable name for the stream, but can contain
-    any text you wish. The values are not unique and may be repeated. Examples:
+    """Stream name.
+
+    Often used as a human-readable name for the stream, but can contain any text you
+    wish. The values are not unique and may be repeated.
+
+    Examples:
 
     - Conference in July
     - Stream #10003
@@ -28,8 +31,10 @@ class Stream(BaseModel):
     """Stream switch between on and off.
 
     This is not an indicator of the status "stream is receiving and it is LIVE", but
-    rather an on/off switch. When stream is switched off, there is no way to process
-    it: PULL is deactivated and PUSH will return an error.
+    rather an on/off switch.
+
+    When stream is switched off, there is no way to process it: PULL is deactivated
+    and PUSH will return an error.
 
     - true – stream can be processed
     - false – stream is off, and cannot be processed
@@ -38,9 +43,12 @@ class Stream(BaseModel):
     auto_record: Optional[bool] = None
     """Enables autotomatic recording of the stream when it started.
 
-    So you don't need to call recording manually. Result of recording is
-    automatically added to video hosting. For details see the
-    /streams/`start_recording` method and in knowledge base Values:
+    So you don't need to call recording manually.
+
+    Result of recording is automatically added to video hosting. For details see the
+    /streams/`start_recording` method and in knowledge base
+
+    Values:
 
     - true – auto recording is enabled
     - false – auto recording is disabled
@@ -49,23 +57,28 @@ class Stream(BaseModel):
     backup_live: Optional[bool] = None
     """
     State of receiving and transcoding master stream from source by backup server if
-    you pushing stream to "`backup_push_url`" or "`backup_push_url_srt`". Displays
-    the backup server status of PUSH method only. For PULL a "live" field is always
-    used, even when origin servers are switched using round robin scheduling (look
-    "uri" field for details).
+    you pushing stream to "`backup_push_url`" or "`backup_push_url_srt`".
+
+    Displays the backup server status of PUSH method only. For PULL a "live" field
+    is always used, even when origin servers are switched using round robin
+    scheduling (look "uri" field for details).
     """
 
     backup_push_url: Optional[str] = None
     """URL to PUSH master stream to our backup server using RTMP/S protocols.
 
-    Servers for the main and backup streams are distributed geographically. Mainly
-    sending one stream to main server is enough. But if you need a backup stream,
-    then this is the field to PUSH it. To use RTMPS just manually change the
-    protocol name from "rtmp://" to "rtmps://". The backup logs are as follows: In
-    PUSH mode, you initiate sending a stream from your machine. If your stream stops
-    or breaks for some reason and it stops coming to the main server, then after
-    3-10 seconds of waiting the stream will turn off or the backup one will be
-    automatically turned on, if you are pushing it too.
+    Servers for the main and backup streams are distributed geographically.
+
+    Mainly sending one stream to main server is enough. But if you need a backup
+    stream, then this is the field to PUSH it.
+
+    To use RTMPS just manually change the protocol name from "rtmp://" to
+    "rtmps://".
+
+    The backup logs are as follows: In PUSH mode, you initiate sending a stream from
+    your machine. If your stream stops or breaks for some reason and it stops coming
+    to the main server, then after 3-10 seconds of waiting the stream will turn off
+    or the backup one will be automatically turned on, if you are pushing it too.
     """
 
     backup_push_url_srt: Optional[str] = None
@@ -105,12 +118,15 @@ class Stream(BaseModel):
     dash_url: Optional[str] = None
     """MPEG-DASH output.
 
-    URL for transcoded result stream in MPEG-DASH format, with .mpd link. Low
-    Latency support: YES. This is CMAF-based MPEG-DASH stream. Encoder and packager
-    dynamically assemble the video stream with fMP4 fragments. Chunks have ±2-4
-    seconds duration depending on the settings. All chunks for DASH are transferred
-    through CDN using chunk transfer technology, which allows to use all the
-    advantages of low latency delivery of DASH.
+    URL for transcoded result stream in MPEG-DASH format, with .mpd link.
+
+    Low Latency support: YES.
+
+    This is CMAF-based MPEG-DASH stream. Encoder and packager dynamically assemble
+    the video stream with fMP4 fragments. Chunks have ±2-4 seconds duration
+    depending on the settings. All chunks for DASH are transferred through CDN using
+    chunk transfer technology, which allows to use all the advantages of low latency
+    delivery of DASH.
 
     - by default low latency is ±4 sec, because it's stable for almost all last-mile
       use cases.
@@ -124,8 +140,11 @@ class Stream(BaseModel):
     """DVR duration in seconds if DVR feature is enabled for the stream.
 
     So this is duration of how far the user can rewind the live stream.
-    `dvr_duration` range is [30...14400]. Maximum value is 4 hours = 14400 seconds.
-    If you need more, ask the Support Team please.
+
+    `dvr_duration` range is [30...14400].
+
+    Maximum value is 4 hours = 14400 seconds. If you need more, ask the Support Team
+    please.
     """
 
     dvr_enabled: Optional[bool] = None
@@ -136,16 +155,17 @@ class Stream(BaseModel):
     """
 
     finished_at_primary: Optional[str] = None
-    """Time when the stream ended for the last time.
-
-    Datetime in ISO 8601. After restarting the stream, this value is not reset to
-    "null", and the time of the last/previous end is always displayed here. That is,
-    when the start time is greater than the end time, it means the current session
-    is still ongoing and the stream has not ended yet. If you want to see all
-    information about acitivity of the stream, you can get it from another method
-    /streaming/statistics/ffprobe. This method shows aggregated activity parameters
-    during a time, when stream was alive and transcoded. Also you can create graphs
-    to see the activity. For example
+    """Time when the stream ended for the last time. Datetime in ISO 8601.
+
+    After restarting the stream, this value is not reset to "null", and the time of
+    the last/previous end is always displayed here. That is, when the start time is
+    greater than the end time, it means the current session is still ongoing and the
+    stream has not ended yet.
+
+    If you want to see all information about acitivity of the stream, you can get it
+    from another method /streaming/statistics/ffprobe. This method shows aggregated
+    activity parameters during a time, when stream was alive and transcoded. Also
+    you can create graphs to see the activity. For example
     /streaming/statistics/ffprobe?interval=6000&`date_from`=2023-10-01&`date_to`=2023-10-11&`stream_id`=12345
     """
 
@@ -156,12 +176,15 @@ class Stream(BaseModel):
     """HLS output.
 
     URL for transcoded result of stream in HLS CMAF format, with .m3u8 link.
-    Recommended for use for all HLS streams. Low Latency support: YES. This is
-    CMAF-based HLS stream. Encoder and packager dynamically assemble the video
-    stream with fMP4 fragments. Chunks have ±2-4 seconds duration depending on the
-    settings. All chunks for LL-HLS are transferred through CDN via dividing into
-    parts (small segments `#EXT-X-PART` of 0.5-1.0 sec duration), which allows to
-    use all the advantages of low latency delivery of LL-HLS.
+    Recommended for use for all HLS streams.
+
+    Low Latency support: YES.
+
+    This is CMAF-based HLS stream. Encoder and packager dynamically assemble the
+    video stream with fMP4 fragments. Chunks have ±2-4 seconds duration depending on
+    the settings. All chunks for LL-HLS are transferred through CDN via dividing
+    into parts (small segments `#EXT-X-PART` of 0.5-1.0 sec duration), which allows
+    to use all the advantages of low latency delivery of LL-HLS.
 
     - by default low latency is ±5 sec, because it's stable for almost all last-mile
       use cases.
@@ -185,12 +208,18 @@ class Stream(BaseModel):
     """HLS output for legacy devices.
 
     URL for transcoded result of stream in HLS MPEG-TS (.ts) format, with .m3u8
-    link. Low Latency support: NO. Some legacy devices or software may require
-    MPEG-TS (.ts) segments as a format for streaming, so we provide this options
-    keeping backward compatibility with any of your existing workflows. For other
-    cases it's better to use "`hls_cmaf_url`" instead. You can use this legacy HLSv6
-    format based on MPEG-TS segmenter in parallel with main HLS CMAF. Both formats
-    are sharing same segments size, manifest length (DVR), etc.
+    link.
+
+    Low Latency support: NO.
+
+    Some legacy devices or software may require MPEG-TS (.ts) segments as a format
+    for streaming, so we provide this options keeping backward compatibility with
+    any of your existing workflows. For other cases it's better to use
+    "`hls_cmaf_url`" instead.
+
+    You can use this legacy HLSv6 format based on MPEG-TS segmenter in parallel with
+    main HLS CMAF. Both formats are sharing same segments size, manifest length
+    (DVR), etc.
 
     It is also possible to use additional modifier-attributes:
 
@@ -200,10 +229,13 @@ class Stream(BaseModel):
       length multiple of whole seconds, or a fractional number separated by a dot
       for chunks that are not multiples of seconds. This attribute allows you to
       determine duration in seconds at the level of analyzing the logs of CDN
-      requests and compare it with file size (so to use it in your analytics). Such
-      modifier attributes are applied manually and added to the link obtained from
-      this field. I.e. `<hls_url>?get_duration_sec=true` Example:
-      `https://demo.gvideo.io/mpegts/2675_19146/master_mpegts.m3u8?get_duration_sec=true`
+      requests and compare it with file size (so to use it in your analytics).
+
+    Such modifier attributes are applied manually and added to the link obtained
+    from this field. I.e. `<hls_url>?get_duration_sec=true`
+
+    Example:
+    `https://demo.gvideo.io/mpegts/2675_19146/master_mpegts.m3u8?get_duration_sec=true`
 
     ```
     #EXTM3U
@@ -230,14 +262,19 @@ class Stream(BaseModel):
     """A URL to a built-in HTML web player with the stream inside.
 
     It can be inserted into an iframe on your website and the video will
-    automatically play in all browsers. Please, remember that transcoded streams
-    from "`hls_cmaf_url`" with .m3u8 at the end, and from "`dash_url`" with .mpd at
-    the end are to be played inside video players only. For example: AVplayer on
-    iOS, Exoplayer on Android, HTML web player in browser, etc. General bowsers like
-    Chrome, Firefox, etc cannot play transcoded streams with .m3u8 and .mpd at the
-    end. The only exception is Safari, which can only play Apple's HLS .m3u8 format
-    with limits. That's why you may need to use this HTML web player. Please, look
-    Knowledge Base for details. Example of usage on a web page:
+    automatically play in all browsers.
+
+    Please, remember that transcoded streams from "`hls_cmaf_url`" with .m3u8 at the
+    end, and from "`dash_url`" with .mpd at the end are to be played inside video
+    players only. For example: AVplayer on iOS, Exoplayer on Android, HTML web
+    player in browser, etc. General bowsers like Chrome, Firefox, etc cannot play
+    transcoded streams with .m3u8 and .mpd at the end. The only exception is Safari,
+    which can only play Apple's HLS .m3u8 format with limits.
+
+    That's why you may need to use this HTML web player. Please, look Knowledge Base
+    for details.
+
+    Example of usage on a web page:
 
     <iframe width="560" height="315" src="https://player.gvideo.co/streams/2675_201693" frameborder="0" allow="autoplay; encrypted-media" allowfullscreen></iframe>
     """
@@ -249,7 +286,9 @@ class Stream(BaseModel):
     """
     Visualization mode for 360° streams, how the stream is rendered in our web
     player ONLY. If you would like to show video 360° in an external video player,
-    then use parameters of that video player. Modes:
+    then use parameters of that video player.
+
+    Modes:
 
     - regular – regular “flat” stream
     - vr360 – display stream in 360° mode
@@ -270,9 +309,11 @@ class Stream(BaseModel):
     """
 
     push_url: Optional[str] = None
-    """
-    URL to PUSH master stream to our main server using RTMP and RTMPS protocols. To
-    use RTMPS just manually change the protocol name from "rtmp://" to "rtmps://".
+    """URL to PUSH master stream to our main server using RTMP and RTMPS protocols.
+
+    To use RTMPS just manually change the protocol name from "rtmp://" to
+    "rtmps://".
+
     Use only 1 protocol of sending a master stream: eitheronly RTMP/S (`push_url`),
     or only SRT (`push_url_srt`).
 
@@ -287,8 +328,9 @@ class Stream(BaseModel):
 
     Please note that 1 connection and 1 protocol can be used at a single moment in
     time per unique stream key input. Trying to send 2+ connection requests into
-    `push_url` to once, or 2+ protocols at once will not lead to a result. For
-    example, transcoding process will fail if:
+    `push_url` to once, or 2+ protocols at once will not lead to a result.
+
+    For example, transcoding process will fail if:
 
     - you are pushing primary and backup RTMP to the same single `push_url`
       simultaneously
@@ -303,31 +345,37 @@ class Stream(BaseModel):
     """
 
     push_url_srt: Optional[str] = None
-    """
-    URL to PUSH master stream to our main server using SRT protocol. Use only 1
-    protocol of sending a master stream: eitheronly RTMP/S (`push_url`), or only SRT
-    (`push_url_srt`).
+    """URL to PUSH master stream to our main server using SRT protocol.
+
+    Use only 1 protocol of sending a master stream: eitheronly RTMP/S (`push_url`),
+    or only SRT (`push_url_srt`).
+
+    **Setup SRT latency on your sender side**
 
-    **Setup SRT latency on your sender side** SRT is designed as a low-latency
-    transport protocol, but real networks are not always stable and in some cases
-    the end-to-end path from the venue to the ingest point can be long. For this
-    reason, it is important to configure the latency parameter carefully to match
-    the actual network conditions. Small latency values may lead to packet loss when
-    jitter or retransmissions occur, while very large values introduce unnecessary
-    end-to-end delay. \\**Incorrect or low default value is one of the most common
-    reasons for packet loss, frames loss, and bad picture.\\**
+    SRT is designed as a low-latency transport protocol, but real networks are not
+    always stable and in some cases the end-to-end path from the venue to the ingest
+    point can be long. For this reason, it is important to configure the latency
+    parameter carefully to match the actual network conditions.
+
+    Small latency values may lead to packet loss when jitter or retransmissions
+    occur, while very large values introduce unnecessary end-to-end delay.
+    \\**Incorrect or low default value is one of the most common reasons for packet
+    loss, frames loss, and bad picture.\\**
 
     We therefore recommend setting latency manually rather than relying on the
     default, to ensure the buffer is correctly sized for your environment. A
     practical range is 400–2000 ms, with the exact value chosen based on RTT,
-    jitter, and expected packet loss. Be sure to check and test SRT settings on your
-    sender side. The default values do not take into account your specific scenarios
-    and do not work well. If necessary, ask us and we will help you.
+    jitter, and expected packet loss.
+
+    Be sure to check and test SRT settings on your sender side. The default values
+    do not take into account your specific scenarios and do not work well. If
+    necessary, ask us and we will help you.
 
     Please note that 1 connection and 1 protocol can be used at a single moment in
     time per unique stream key input. Trying to send 2+ connection requests into
-    `push_url_srt` to once, or 2+ protocols at once will not lead to a result. For
-    example, transcoding process will fail if:
+    `push_url_srt` to once, or 2+ protocols at once will not lead to a result.
+
+    For example, transcoding process will fail if:
 
     - you are pushing primary and backup SRT to the same single `push_url_srt`
       simultaneously
@@ -340,37 +388,52 @@ class Stream(BaseModel):
     push_url_whip: Optional[str] = None
     """URL to PUSH WebRTC stream to our server using WHIP protocol.
 
-    **WebRTC WHIP to LL-HLS and DASH** Video Streaming supports WebRTC HTTP Ingest
-    Protocol (WHIP), and WebRTC to HLS/DASH converter. As a result you can stream
-    from web broswers natively. **WebRTC WHIP server** We have dedicated WebRTC WHIP
-    servers in our infrastructure. WebRTC WHIP server organizes both signaling and
-    receives video data. Signaling is a term to describe communication between
-    WebRTC endpoints, needed to initiate and maintain a session. WHIP is an open
-    specification for a simple signaling protocol for starting WebRTC sessions in an
-    outgoing direction, (i.e., streaming from your device). There is the primary
-    link only for WHIP, so no backup link. **WebRTC stream encoding parameters** At
-    least one video and audio track both must be present in the stream:
+    **WebRTC WHIP to LL-HLS and DASH**
+
+    Video Streaming supports WebRTC HTTP Ingest Protocol (WHIP), and WebRTC to
+    HLS/DASH converter. As a result you can stream from web broswers natively.
+
+    **WebRTC WHIP server**
+
+    We have dedicated WebRTC WHIP servers in our infrastructure. WebRTC WHIP server
+    organizes both signaling and receives video data. Signaling is a term to
+    describe communication between WebRTC endpoints, needed to initiate and maintain
+    a session. WHIP is an open specification for a simple signaling protocol for
+    starting WebRTC sessions in an outgoing direction, (i.e., streaming from your
+    device).
+
+    There is the primary link only for WHIP, so no backup link.
+
+    **WebRTC stream encoding parameters**
+
+    At least one video and audio track both must be present in the stream:
 
     - Video must be encoded with H.264.
-    - Audio must be encoded with OPUS. Note. Specifically for WebRTC mode a method
-      of constant transcoding with an initial given resolution is used. This means
-      that if WebRTC in the end-user's browser decides to reduce the quality or
-      resolution of the master stream (to let say 360p) due to restrictions on the
-      end-user's device (network conditions, CPU consumption, etc.), the transcoder
-      will still continue to transcode the reduced stream to the initial resolution
-      (let say 1080p ABR). When the restrictions on the end-user's device are
-      removed, quiality will improve again. **WebRTC WHIP Client** We provide a
-      convenient WebRTC WHIP library for working in browsers. You can use our
-      library, or any other you prefer. Simple example of usage is here:
-      https://stackblitz.com/edit/stackblitz-starters-j2r9ar?file=index.html Also
-      try to use the feature in UI of the Customer Portal. In the Streaming section
-      inside the settings of a specific live stream, a new section "Quick start in
-      browser" has been added.
+    - Audio must be encoded with OPUS.
+
+    Note. Specifically for WebRTC mode a method of constant transcoding with an
+    initial given resolution is used. This means that if WebRTC in the end-user's
+    browser decides to reduce the quality or resolution of the master stream (to let
+    say 360p) due to restrictions on the end-user's device (network conditions, CPU
+    consumption, etc.), the transcoder will still continue to transcode the reduced
+    stream to the initial resolution (let say 1080p ABR). When the restrictions on
+    the end-user's device are removed, quiality will improve again.
+
+    **WebRTC WHIP Client**
+
+    We provide a convenient WebRTC WHIP library for working in browsers. You can use
+    our library, or any other you prefer. Simple example of usage is here:
+    https://stackblitz.com/edit/stackblitz-starters-j2r9ar?file=index.html
+
+    Also try to use the feature in UI of the Customer Portal. In the Streaming
+    section inside the settings of a specific live stream, a new section "Quick
+    start in browser" has been added.
 
     Please note that 1 connection and 1 protocol can be used at a single moment in
     time per unique stream key input. Trying to send 2+ connection requests into
-    `push_url_whip` to once, or 2+ protocols at once will not lead to a result. For
-    example, transcoding process will fail if:
+    `push_url_whip` to once, or 2+ protocols at once will not lead to a result.
+
+    For example, transcoding process will fail if:
 
     - you are pushing primary and backup WHIP to the same single `push_url_whip`
       simultaneously
@@ -389,7 +452,9 @@ class Stream(BaseModel):
     """Method of recording a stream.
 
     Specifies the source from which the stream will be recorded: original or
-    transcoded. Types:
+    transcoded.
+
+    Types:
 
     - "origin" – To record RMTP/SRT/etc original clean media source.
     - "transcoded" – To record the output transcoded version of the stream,
@@ -403,10 +468,12 @@ class Stream(BaseModel):
     """
     An instant screenshot taken from a live stream, and available as a static JPEG
     image. Resolution 1080 pixels wide, or less if the original stream has a lower
-    resolution. Screenshot is taken every 10 seconds while the stream is live. This
-    field contains a link to the last screenshot created by the system. Screenshot
-    history is not stored, so if you need a series of screenshots over time, then
-    download them.
+    resolution.
+
+    Screenshot is taken every 10 seconds while the stream is live. This field
+    contains a link to the last screenshot created by the system. Screenshot history
+    is not stored, so if you need a series of screenshots over time, then download
+    them.
     """
 
     started_at_backup: Optional[str] = None
@@ -418,36 +485,40 @@ class Stream(BaseModel):
     started_at_primary: Optional[str] = None
     """Time of the last session when main server started receiving the stream.
 
-    Datetime in ISO 8601. This means that if the stream was started 1 time, then
-    here will be the time it was started. If the stream was started several times,
-    or restarted on your side, then only the time of the last session is displayed
-    here.
+    Datetime in ISO 8601.
+
+    This means that if the stream was started 1 time, then here will be the time it
+    was started. If the stream was started several times, or restarted on your side,
+    then only the time of the last session is displayed here.
     """
 
     transcoded_qualities: Optional[List[str]] = None
     """Array of qualities to which live stream is transcoded"""
 
     transcoding_speed: Optional[float] = None
-    """Speed of transcoding the stream. Mainly it must be 1.0 for real-time processing.
+    """Speed of transcoding the stream.
 
-    May be less than 1.0 if your stream has problems in delivery due to your local
-    internet provider's conditions, or the stream does not meet stream inbound
-    requirements. See Knowledge Base for details.
+    Mainly it must be 1.0 for real-time processing. May be less than 1.0 if your
+    stream has problems in delivery due to your local internet provider's
+    conditions, or the stream does not meet stream inbound requirements. See
+    Knowledge Base for details.
     """
 
     uri: Optional[str] = None
-    """
-    When using PULL method, this is the URL to pull a stream from. You can specify
-    multiple addresses separated by a space (" "), so you can organize a backup
-    plan. In this case, the specified addresses will be selected one by one using
-    round robin scheduling. If the first address does not respond, then the next one
-    in the list will be automatically requested, returning to the first and so on in
-    a circle. Also, if the sucessfully working stream stops sending data, then the
-    next one will be selected according to the same scheme. After 2 hours of
-    inactivity of your original stream, the system stops PULL requests and the
-    stream is deactivated (the "active" field switches to "false"). Please, note
-    that this field is for PULL only, so is not suitable for PUSH. Look at fields
-    "`push_url`" and "`push_url_srt`" from GET method.
+    """When using PULL method, this is the URL to pull a stream from.
+
+    You can specify multiple addresses separated by a space (" "), so you can
+    organize a backup plan. In this case, the specified addresses will be selected
+    one by one using round robin scheduling. If the first address does not respond,
+    then the next one in the list will be automatically requested, returning to the
+    first and so on in a circle. Also, if the sucessfully working stream stops
+    sending data, then the next one will be selected according to the same scheme.
+
+    After 2 hours of inactivity of your original stream, the system stops PULL
+    requests and the stream is deactivated (the "active" field switches to "false").
+
+    Please, note that this field is for PULL only, so is not suitable for PUSH. Look
+    at fields "`push_url`" and "`push_url_srt`" from GET method.
     """
 
     video_height: Optional[float] = None

gcore/types/streaming/stream_create_clip_params.py

@@ -9,40 +9,51 @@ __all__ = ["StreamCreateClipParams"]
 
 class StreamCreateClipParams(TypedDict, total=False):
     duration: Required[int]
-    """
-    Requested segment duration in seconds to be cut. Please, note that cutting is
-    based on the idea of instantly creating a clip, instead of precise timing. So
-    final segment may be:
+    """Requested segment duration in seconds to be cut.
+
+    Please, note that cutting is based on the idea of instantly creating a clip,
+    instead of precise timing. So final segment may be:
 
     - Less than the specified value if there is less data in the DVR than the
       requested segment.
     - Greater than the specified value, because segment is aligned to the first and
       last key frames of already stored fragment in DVR, this way -1 and +1 chunks
-      can be added to left and right. Duration of cutted segment cannot be greater
-      than DVR duration for this stream. Therefore, to change the maximum, use
-      "`dvr_duration`" parameter of this stream.
+      can be added to left and right.
+
+    Duration of cutted segment cannot be greater than DVR duration for this stream.
+    Therefore, to change the maximum, use "`dvr_duration`" parameter of this stream.
     """
 
     expiration: int
-    """
-    Expire time of the clip via a public link. Unix timestamp in seconds, absolute
-    value. This is the time how long the instant clip will be stored in the server
-    memory and can be accessed via public HLS/MP4 links. Download and/or use the
-    instant clip before this time expires. After the time has expired, the clip is
-    deleted from memory and is no longer available via the link. You need to create
-    a new segment, or use `vod_required: true` attribute. If value is omitted, then
-    expiration is counted as +3600 seconds (1 hour) to the end of the clip (i.e.
-    `unix timestamp = <start> + <duration> + 3600`). Allowed range: 1m <= expiration
-    <= 4h. Example:
+    """Expire time of the clip via a public link.
+
+    Unix timestamp in seconds, absolute value.
+
+    This is the time how long the instant clip will be stored in the server memory
+    and can be accessed via public HLS/MP4 links. Download and/or use the instant
+    clip before this time expires.
+
+    After the time has expired, the clip is deleted from memory and is no longer
+    available via the link. You need to create a new segment, or use
+    `vod_required: true` attribute.
+
+    If value is omitted, then expiration is counted as +3600 seconds (1 hour) to the
+    end of the clip (i.e. `unix timestamp = <start> + <duration> + 3600`).
+
+    Allowed range: 1m <= expiration <= 4h.
+
+    Example:
     `24.05.2024 14:00:00 (GMT) + 60 seconds of duration + 3600 seconds of expiration = 24.05.2024 15:01:00 (GMT) is Unix timestamp = 1716562860`
     """
 
     start: int
-    """
-    Starting point of the segment to cut. Unix timestamp in seconds, absolute value.
-    Example: `24.05.2024 14:00:00 (GMT) is Unix timestamp = 1716559200` If a value
-    from the past is specified, it is used as the starting point for the segment to
-    cut. If the value is omitted, then clip will start from now.
+    """Starting point of the segment to cut.
+
+    Unix timestamp in seconds, absolute value. Example:
+    `24.05.2024 14:00:00 (GMT) is Unix timestamp = 1716559200`
+
+    If a value from the past is specified, it is used as the starting point for the
+    segment to cut. If the value is omitted, then clip will start from now.
     """
 
     vod_required: bool