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Variables

CODEC_FORMAT: string
CODEC_ID_ANY: number
CODEC_ID_DISABLE: number
RTP_HEADER_EXTENSION_FORMAT: string

Functions

  • Reads the content of a #GKeyFile of the following format into a #GList of #FsCodec structures.

    Example: |[ [audio/codec1] clock-rate=8000

    [audio/codec1:1] clock-rate=16000

    [audio/codec2] one_param=QCIF another_param=WOW

    [video/codec3] wierd_param=42 feedback:nack/pli=1 feedback:tfrc=


    @param filename Name of the #GKeyFile to read the codecs parameters from

    Parameters

    • filename: string

    Returns Farstream.Codec[]

  • rtp_header_extension_list_copy(extensions: any[]): any[]
  • Does a deep copy of a #GList of #FsRtpHeaderExtension

    Parameters

    • extensions: any[]

      a #GList of #FsRtpHeaderExtension

    Returns any[]

  • rtp_header_extension_list_from_keyfile(filename: string, media_type: Farstream.MediaType): any[]
  • Reads the content of a #GKeyFile of the following format into a #GList of #FsRtpHeaderExtension structures.

    The groups have a format "rtp-hdrext:audio:XXX" or "rtp-hdrext:video:XXX" where XXX is a unique string (per media type).

    The valid keys are: id: a int between in the 1-255 and 4096-4351 ranges uri: a URI describing the RTP Header Extension direction (optional): To only send or receive a RTP Header Extension, possible values are "send", "receive", "none" or "both". Defaults to "both"

    Example: |[ [rtp-hdrext:audio:a] id=1 uri=urn:ietf:params:rtp-hdrext:toffset

    [rtp-hdrext:audio:abc] id=3 uri=urn:ietf:params:rtp-hdrext:ntp-64 direction=receive


    @param filename Name of the #GKeyFile to read the RTP Header Extensions from
    @param media_type The media type for which to get header extensions

    Parameters

    Returns any[]

  • These default codec preferences should work with the elements that are available in the main GStreamer element repositories. They should be suitable for standards based protocols like SIP or XMPP.

    Parameters

    • element: Gst.Element

      Element for which to fetch default codec preferences

    Returns Farstream.Codec[]

  • These default rtp header extension preferences should work with the elements that are available in the main GStreamer element repositories. They should be suitable for standards based protocols like SIP or XMPP.

    Parameters

    • element: Gst.Element

      Element for which to fetch default RTP Header Extension preferences

    • media_type: Farstream.MediaType

      The #FsMediaType for which to get default RTP Header Extension preferences

    Returns Farstream.Codec[]

  • utils_set_bitrate(element: Gst.Element, bitrate: number): void
  • This allows setting the bitrate on all elements that have a "bitrate" property without having to know the type or of the unit used by that element.

    This will be obsolete in 0.11 (when all elements use bit/sec for the "bitrate" property.

    Parameters

    • element: Gst.Element

      The #GstElement

    • bitrate: number

      The bitrate in bits/sec

    Returns void

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