Stereo Tool

Hi All,

Running ST 7.60 x64 and working OK.

I was wondering how people setup buffers with sound card input and FM Output.

At the moment I have configured my input (VAC @192KHz) for 3 seconds and the FM Output (192KHz sound card) for the same with resample quality 7 which appears to keep up well.

Previously I ran 5 seconds for both input and FM Output but the buffer expires over time, sometimes really quickly, other times it runs for days and is fine.

It seems to me that running a larger buffer requires more CPU and has a higher chance of getting out of sync, then dieing before restarting again.

I would like to know what other people are using for the buffer settings on ST.

Many thanks

BeDazzler.

Since you dont use asio it is a must to use "sync" Buffer can be as low as how much your CPU can handle. So for non-asio easy 100ms with decent CPU.

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In the Help file Hans says not to use the Input Buffer:

Input Buffer size
Do not use this, leave at 0.

C.

Thanks for the replies.

Apologies, I should have clarified my FM Output sound card is using ASIO.

Do I need to use "sync output" in this case ?

I thought it did not apply to ASIO output.



BeDazzler.

It does if you use a different sound card for input. I'm not 100% sure if it work properly with ASIO (it should but I vaguely recall hearing something about that).

Hi Hans,

My input sound source is VAC @ 192KHz which I assume is treated like another sound card ?

The input source itself is 44.1 which plays into a 192KHz VAC then into ST.

I have been able to get this working by setting both the input buffer and the FM Output buffer to zero, however I still sometimes hear issues on our test rig where the audio breaks up, disappears for 2 seconds and then comes back.

Perhaps I need to run with a 2 second buffer on both the input and FM Output to give ST a little breathing space.

Happy to hear suggestions.

Many thanks

BeDazzler.

Synchronize won't work with a 0 second buffer! So you need more (2 seconds would be good).

By the way, playing 44.1 into 192 kHz VAC will give you resampling artifacts... if possible switch the whole thing to 176.4 kHz.

Hi Hans,

Thanks for the reply.

The playout app only provides output in either 44.1 or 48KHz (there are no other options). Currently set @ 44.1KHz as this is the format of all audio playing (flac and wav files).

Are you saying I should play out of the audio software @ 44.1KHz into a VAC @ 44.1KHz which then feeds ST.
Then set 'synchronise output' in FM Output and let ST handle the resync between VAC output and the FM soundcard output ?

I have a 2 second buffer set in the FM Output section. Do I need a buffer set on the input section ? (currently set at 0).

The resampling quality is set to 8 as this is the number where the buffer can keep up.

Can you explain a little about how the resamplng quality works ?

What do the numbers "M: 0.9670" and "x: 0.9998" mean ?

I have not previously tried changing the sample rates on the audio output into VAC, however I have just changed that now and it sounds OK. I don't hear anything unusual and the audio is clear over FM.

BeDazzler.

Hi,

No, but resampling from 44.1 to 176.4 (exactly 4 times 44.1) should give no artifacts, while resampling from 44.1 to 192 (basically 4x48) does.

The numbers:
M is how full the buffer is (should be 1, can range between 0 and 2)
x is how much the sample rate is adjusted at this moment

Thanks Hans, that is a big help.

I have set the input rates all @ 44.1KHz, then changed the ST FM Output and soundcard hardware to 176.4KHz.

Then I have dragged the FM Output buffer to 5 seconds to obtain M: 0.9800 and x: 0.999 which works well and ST is keeping up.

CPU load is 30% across 8 cores, ST processing load is 52%.

With Composite Clipper enabled (CPU Strictness 5) M: 0.9770 and not growing much past that number, so I imagine over time the buffer will decrease and cause problems. It has fallen to 0.9740 in the last 20 mins.

Is there anything else I can do to help the composite clipper work better?

BeDazzler.