<div>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.<br>
</div>
<div>- Dake<br></div>
<div class="gmail_quote">2010/6/9 Wim Taymans <span dir="ltr"><<a href="mailto:wim.taymans@gmail.com">wim.taymans@gmail.com</a>></span><br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">
<div class="im">On Wed, 2010-06-09 at 17:31 -0700, Dake Gu wrote:<br>> I am investigating an a/v sync issue of h264 + aac-LATM rtsp<br>> streaming.<br>><br>> The rtpmp4adepay retrieves 32bits rtptime from buffer header and<br>
> passes rtptime to gst_base_rtp_depayload_push_ts()<br>> According to the doc: child class might override set_gst_timestamp to<br>> calculate GST timestamp from RTP timestamp.<br>> /* non-pure function used to convert from RTP timestamp to GST timestamp<br>
> * this function is used by the child class before gst_pad_pushing */<br>><br>> void (*set_gst_timestamp) (GstBaseRTPDepayload *filter, guint32 timestamp, GstBuffer *buf);<br>> But in fact, none of the child depay class overrides the function,<br>
> instead, they are using gstbuffer timestamp from upstream.<br>><br>> My question is:<br>> - How is the gst buffer timestamp generated and pass to rtpdepay?<br><br></div>It's the timestamp generated by the live capture element (like udpsrc)<br>
<br>> - Why is rtptime not used to generate a gst buffer timestamp?<br><br>The rtptime is used for generating buffer timestamps in the<br>jitterbuffer, not in the depayloader. The depayloader simply copies the<br>incomming gstreamer timestamp to the outgoing buffers.<br>
<br>Wim<br><br>><br>> Thanks in advance!<br>><br>> - Dake<br>><br>> ------------------------------------------------------------------------------<br>> ThinkGeek and WIRED's GeekDad team up for the Ultimate<br>
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