[gst-devel] question about rtptime in depayload

[Dake Gu]

If I understand correctly, updsrc generates buffer timestamp from running
clock,  and jitter buffer uses sender's current rtptime, base rtptime to
calculate receiver skew and adjust the buffer timestamp.  Thanks.
- Dake
2010/6/9 Wim Taymans <wim.taymans at gmail.com>

> 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
> >
> >
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