[gst-devel] question about rtptime in depayload

Wim Taymans wim.taymans at gmail.com
Thu Jun 10 08:54:26 CEST 2010

On Wed, 2010-06-09 at 17:31 -0700, Dake Gu wrote:
> I am investigating an a/v sync issue of h264 + aac-LATM rtsp
> streaming.
> The rtpmp4adepay retrieves 32bits rtptime from buffer header and
> passes rtptime to gst_base_rtp_depayload_push_ts()
> According to the doc:  child class might override set_gst_timestamp to
> calculate GST timestamp from RTP timestamp.
> /* non-pure function used to convert from RTP timestamp to GST timestamp
>    * this function is used by the child class before gst_pad_pushing */
>   void (*set_gst_timestamp) (GstBaseRTPDepayload *filter, guint32 timestamp, GstBuffer *buf);
> But in fact,  none of the child depay class overrides the function,
> instead, they are using gstbuffer timestamp from upstream.
> My question is:
> -  How is the gst buffer timestamp generated and pass to rtpdepay?

It's the timestamp generated by the live capture element (like udpsrc)

> -  Why is rtptime not used to generate a gst buffer timestamp?

The rtptime is used for generating buffer timestamps in the
jitterbuffer, not in the depayloader. The depayloader simply copies the
incomming gstreamer timestamp to the outgoing buffers.


> Thanks in advance!
> - Dake
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