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rtpbasepayload: Improve documentation for perfect-rtptime

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Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719383
This commit is contained in:
Sebastian Rasmussen 2014-01-22 17:47:02 +01:00 committed by Wim Taymans
parent 713dfe0d70
commit 865a5d1c8f
1 changed files with 15 additions and 4 deletions
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19
gst-libs/gst/rtp/gstrtpbasepayload.c
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@ -224,10 +224,21 @@ gst_rtp_base_payload_class_init (GstRTPBasePayloadClass * klass)
/**
* GstRTPBasePayload:perfect-rtptime:
*
* Try to use the offset fields to generate perfect RTP timestamps. when this
* option is disabled, RTP timestamps are generated from the GStreamer
* timestamps, which could result in RTP timestamps that don't increment with
* the amount of data in the packet.
* 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.
*/
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_PERFECT_RTPTIME,
g_param_spec_boolean ("perfect-rtptime", "Perfect RTP Time",