AES sender pipeline options

[Sebastian Dröge]

Hi James,

On Thu, 2019-01-24 at 13:06 -0600, James Cowdery wrote:
> I've put together an audio pipeline to stream a wav file that will be readily
> accepted by AES67 compatible receivers e.g. Ravenna or Dante.

That's great to hear. When implementing all those things I focussed
mostly on GStreamer-based receivers but in theory this should also work
for senders, and a colleague of mine also got it to work with Dante a
while ago.

>  I also send out the SDP using a SAP packet separately.

How does your SDP look like exactly? Most importantly, does it signal
the PTP clock correctly and the media clock offset (i.e. the diff
between PTP and RTP timestamps)?

gst-rtsp-server can produce RFC7273 (and thus AES67) compatible SDPs,
so maybe you can take some inspiration from that code.

> 
> [...]             
> 
> This works fine for receivers that are not ptp locked e.g. ffplay but my
> Ravenna receiver complains about the RTP timestamps being too late or not
> synchronized to PTP.

Can you share the exact errors you get? But my guess would be the media
clock offset in the SDP being wrong.

> I've created the PTP clock which appears to lock to my PTP grandmaster OK
> and I execute gst_pipeline_use_clock to lock the pipeline to the PTP clock
> once it's synched. However, I think I need to set a few options to get the
> timestamps correct. The options I'm unsure about are:
> 
> rtpbin.buffer-mode
> rtpbin.ntp-time-source
> pipeline latency (gst_pipeline_set_latency)

These ones don't matter at all for senders.

> rtpL16pay.timestamp-offset

This is the "start" RTP timestamp to be used by outgoing packets, and
by default is randomized. You can make use of this for creating the
SDP.

In addition, the max/min-ptime properties on the payloader might be
relevant. AES67 wants specific packet sizes.

-- 
Sebastian Dröge, Centricular Ltd · https://www.centricular.com

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