Thanks Marcus,

It sounds like a good idea but I think I haven't explained my flow graph
well enough. This a simplification of what I've got.

[image: Inline images 1]
I've taken away one of the cards, filters and others. Can I still use your
approach using "streams to vector" -> "vector to streams" in this
situation?

Thanks,
Murray

On 13 January 2016 at 10:10, Marcus Müller <[email protected]> wrote:

> Well, that could well be; at 96kHz, 1s after a day is a drift of
> [image: $\frac{\SI{1}{\second}}{24\cdot 60 \cdot
> 60\si{\second}}=11.57\,\mathrm{ppm}$]
> which would be a relatively good value for soundcard oscillators.
>
> For drifts that small, you probably won't be able to correct this from
> signal.
>
> I'd say, since you can live with dropping samples, do the following:
>
> source 1   ->                                       ->
> ...        ->streams to vector -> vector to streams ->
> source N   ->                                       ->
>
>
> that enforces sample-synchronity. You then set a maximum size for the
> output buffers of the source blocks, if you'd rather lose few samples then
> seeing a growing delay and then lose a bigger bunch of samples.
>
> Best regards,
> Marcus
>
>
> On 01/13/2016 08:30 AM, Murray Thomson wrote:
>
> Hi,
>
> I'm using three audio cards in my flow graph. The internal and a USB audio
> cards run at 96 kHz and I use a virtual audio card with 5 subdevices
> loading the snd_aloop kernel module that runs at 48 kHz.
>
> I've noticed that after some days of running there seems to be a delay
> (around 1 second) processing the signal.
>
> I was wondering if this could be due to the differences in the clocks of
> the audio cards. Is there a way to correct these differences? I can afford
> losing data every now and again, but an increasing delay it's a problem for
> me.
>
> Thanks,
> Murray
>
>
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