Generating and presenting sound stimuli with Spike2 from Analogue output channels

[pichem]

Hi all,

I read a few posts on the topic but I cannot find exactly what I want to know... I would like to get some advice about the best way of doing what I want to do. I want to use the graphical sequence editor to generate pure tones to be presented through earphones during my experiment. I know I to output the sound stimuli but I don't know if I could use that output to connect my earphones directly to the analogue outputs on the Power1401. I need two channels because I will present sounds on the right or the left depending on the condition. I wonder if I can use an adaptor to "split" my phone jack connector (from the earphones) to simply receive one analogue output channel on the right and one on the left... Would the sound intensity be loud enough or I need some kind of audio amplifier between the 1401 and my earphones? I will measure evokes potentials so the timing is important and I am worried that I will have some jitter if I use my sound card or E-prime.

Thanks

Mathieu

[Pawel Kusmierek]

A veteran of sound playback from Spike2 here

You will be definitely better off with a headphone amplifier, I don't think that 1401 outputs will work well directly with headphones. Headphones are relatively low impedance.

Presonus HP4 is a good piece of hardware, and its maximum input level is +18 dBu so it will not overload with 1401's -5..+5V range.

Also it has separate inputs for L & R channel, which will simplify connecting to the 1401, just use two BNC(F) to TS(M) adapters lke this https://www.pomonaelectronics.com/produ ... phone-plug and two BNC cables to connect two 1401's DACs to the amp.

If you are going to use always only one channnel at a time, there is also an option to sticking to one DAC and using a switch controlled via 1401's digital I/O, but there is a risk of clicks.

Using a 1401 will ensure perfect timing for ERP, but keep in mind that audio signal quality will be not as good as from a high-quality sounds cards. 1401 DACs are 16-bit and not optimized for audio, while good sound cards have 24-bit DACs with S/N exceeding 100 dB. This probably won't be an issue if your signals are high-amplitude in digital domain, but I would very careful with low-digital-level signals. If you need low levels, it's probably better to use an analog attaenuator like CED 3505. You would need two of those (one per channel), unless you go with one DAC and a switch.

Also, 1401s DACs have no antialiasing low-pass filters so you may consider adding one, set to 0.5 * your sampling frequency. Again, one option comes from CED (4301). And again, two needed unless using one DAC and a switch.

There is another caveat when using the 1401 to play audio, but this applies to arbitrary (sampled) waveforms, so you should be fine as you are about to generate sine waves within the system. But just in case you wanted to used arbitrary waveforms, like natural sounds or even tones generated externally, saved as wave files, and imported into Spike2/1401: the 1401 cannot exactly replicate sampling frequencies used in standard digital audio (like 44.1 kHz, 48 kHz, etc). So you have to be aware of that and compensate if necessary (but the difference is tiny, so it may be negligible depending on your needs)

[pichem]

Hi Pawel,

I realize that I did not reply to you! Sorry about that! I followed your suggestions and we are now setting up the experiment.

Thanks a lot!

Mathieu

[Pawel Kusmierek]

No problem, I am glad that it was (hopefully) helpful