Apache Labs Community Forum

Hello;

I finally made the jump to protocol 2 Firmware and Thetis 2.6.9

I noticed the audio set up tabs are different in Thetis vs openHPSDR.

In openHPSDR there were three tabs under the audio set up:

Primary - VAC 1 - VAC 2

In the primary tab, you could set the buffer and sample rate for the input.

In Thetis 2.6.9 there are only tabs for VAC 1 and VAC 2

How does one set the buffer size and sample rate for the primary audio?

The goal is to be able to run as low a buffer size as possible to eliminate as much latency in the monitor as possible.

The old version I was running had a buffer of 64 and 128 (many other larger ones too)

I was never able to go below 128 on the buffer without getting audio glitches.

Hopefully I have been clear enough - if not, please let me know what other pertinent information I could add.

Thanks

screenshots attached

That setting no longer exists and you do not need to worry about it.

Also, it was not an "input" setting, but rather a buffer associated with the transfer of IF data between the hardware and the software.

w-u-2-o wrote:That setting no longer exists and you do not need to worry about it.

Also, it was not an "input" setting, but rather a buffer associated with the transfer of IF data between the hardware and the software.

Thanks WU2O;

Do you happen to know if there is a buffer set for that IF data between the hardware and software?
I still notice some slight latency when keying the radio and listening to my audio in the monitor.
That is the issue I am trying to solve as it makes me stumble on words when there is latency.

Thanks - I have viewed and followed the other tutorials you have posted - I greatly appreciate them.

Rich in Seattle - W7VTX

Rich are you able to run an Asio driver instead of MME ?

Rich,

There is very little latency in the IF data buffering. Just a few milliseconds. You have quite a ways to go to optimize your VAC buffer sizes. Select "manual" for all ringbuffers. Try starting with both PortAudios at zero. Then start shaving away at the RingBuffers until audio quality gets bad.

Tony's question is also a good one.

Also, if you are using a lot of audio processing, e.g. the leveler, EQ, CFC, etc., each one of those audio processing stages adds latency.

Unfortunately the monitor function cannot be set to pure unprocessed sidetone, i.e. picked off prior to any audio processing. This would be a wonderful feature to have.

Also, the monitor function comes after the PureSignal predistortion stage, so anything you hear has been predistorted and may sound funny. Thus it is not good for adjusting your audio quality unless PureSignal is disabled.

73,

Scott