Minimum duration of the packet data in ns (can't go above MTU)
Make the payloader timestamp packets according to the Rate-Control=no behaviour specified in the ONVIF replay spec.
The parent of the object. Please note, that when changing the 'parent' property, we don't emit #GObject::notify and #GstObject::deep-notify signals due to locking issues. In some cases one can use #GstBin::element-added or #GstBin::element-removed signals on the parent to achieve a similar effect.
Try to use the offset fields to generate perfect RTP timestamps. When this option is disabled, RTP timestamps are generated from GST_BUFFER_PTS of each payloaded buffer. The PTSes of buffers may not necessarily increment with the amount of data in each input buffer, consider e.g. the case where the buffer arrives from a network which means that the PTS is unrelated to the amount of data. Because the RTP timestamps are generated from GST_BUFFER_PTS this can result in RTP timestamps that also don't increment with the amount of data in the payloaded packet. To circumvent this it is possible to set the perfect rtptime option enabled. When this option is enabled the payloader will increment the RTP timestamps based on GST_BUFFER_OFFSET which relates to the amount of data in each packet rather than the GST_BUFFER_PTS of each buffer and therefore the RTP timestamps will more closely correlate with the amount of data in each buffer. Currently GstRTPBasePayload is limited to handling perfect RTP timestamps for audio streams.
Force buffers to be multiples of this duration in ns (0 disables)
Make the RTP packets' timestamps be scaled with the segment's rate (corresponding to RTSP speed parameter). Disabling this property means the timestamps will not be affected by the set delivery speed (RTSP speed).
Example: A server wants to allow streaming a recorded video in double speed but still have the timestamps correspond to the position in the video. This is achieved by the client setting RTSP Speed to 2 while the server has this property disabled.
Enable writing the CSRC field in allocated RTP header based on RTP source information found in the input buffer's #GstRTPSourceMeta.
If enabled, the payloader will automatically try to enable all the RTP header extensions provided in the src caps, saving the application the need to handle these extensions manually using the GstRTPBasePayload::request-extension: signal.