On Wed, 21 Dec 2005, Zeeshan Ali wrote:
While writing the element, i came accross a problem:
basertpdepayload uses a variable 'clock-rate' for deciding the amount
of time a packet remains in the buffer/queue. This variable is
codec-dependent and it's value is provided by the elements inheriting
from basertpdepayload.
This problem was further discussed at #farsight today, and the conclusion
was that we need to provide this information already at rtprecv.
Currently, information available from RTP headers (timestamp,
payload-type, etc) is parsed by gstrtpbuffer.h and stored for each buffer.
But some information needed in processing RTP packets cannot be found out
from the headers.
This information includes:
- mapping of payload-types (RTP-header field) to codecs (note: the
fixed PTs 0-95 in RFC3551 are an exception)
- clock rate and channel count
- codec parameters
Some of this information can be collected at depay/decoding time, but not
in all cases. For example, for L8, there are no limits for clock-rate, nor
channel count, nor payload-type. This information is however needed by
numerous gst elements: rtpjitterbuf (clock-rate), depay/decoder (codec
params), and the sink (clock/sample-rate).
So the consensus was (me, rob, zeeshan), that application should provide a
"pt<->caps" table to rtpbin (and to rtprecv), which allows rtprecv to
attach this out-of-band agreed information to buffers it pushes out.
Notes:
- spec for how to pass this information in caps:
gst-plugins-good-cvs/gst/rtp/README
Open issues:
- how to manage conflicts - clock-rate set by the app to 44100,
but pipeline has elements which are fixed to 8000... what
happens?
PS Yep, once the ticket system is back up, this has to be added there...
--
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