Re: [LAD] AoIP question

From: Len Ovens <len@email-addr-hidden>
Date: Sun Oct 05 2014 - 23:51:36 EEST

On Sun, 5 Oct 2014, tom@email-addr-hidden wrote:

> Hi, i'm following the thread of the ongoing topics around transmitting
> audio over IP infrastructure.
> As a scenario, at point a) an analog signal is injected that will be
> played back (analog) at point b) with the lowest possible (and constant)
> latency.
> How do you intend to handle diverging clocks of the audio interfaces
> (ADC/DAC) at both (a/b) ends?

AES67 (and other formats) use PTP (IEEE 1588-2008) to keep the system
clocks aligned at a usec level. The media clock is then derived from that.
The media clock on both systems should therefore be syncronis.

BTW, AES67 reading and understanding is not fun or easy as AES67
constantly refers to other documents. Everything in there is already
specified in some other place. AES67 basically just tells which of the
available network standards should be used together so that systems can
talk to each other.

--
Len Ovens
www.ovenwerks.net
_______________________________________________
Linux-audio-dev mailing list
Linux-audio-dev@email-addr-hidden
http://lists.linuxaudio.org/listinfo/linux-audio-dev
Received on Mon Oct 6 00:15:05 2014

This archive was generated by hypermail 2.1.8 : Mon Oct 06 2014 - 00:15:05 EEST