[gst-devel] question about rtptime in depayload

[Wim Taymans]

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.

Wim

> 
> Thanks in advance!
> 
> - Dake
> 
> ------------------------------------------------------------------------------
> ThinkGeek and WIRED's GeekDad team up for the Ultimate 
> GeekDad Father's Day Giveaway. ONE MASSIVE PRIZE to the 
> lucky parental unit.  See the prize list and enter to win: 
> http://p.sf.net/sfu/thinkgeek-promo
> _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel