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Main :: ToneCore Development Kit

Input-output audio latency?
by audiofanatic on 2011-06-11 05:49:47


I'm interested in using the device for an unusual application which requires the audio input-output latency (as measured with an input pulse train and looking at the input and output on a scope and comparing the delay) of 1ms or lower. Is there any chance that the development hardware can do this?

Normally sigma-delta modulators and internal buffering will cause the latency to be longer than 1ms, but I thought since this was such a cool platform and musicians like to have really low input-output audio latency, it might actually be this low. (Otherwise, on most similar platforms, AFAIK the latency is usually just speced to be less than 10ms, which will not suffice for my application.)

Any input from the Line 6 developers would be highly, highly valued.



Re: Input-output audio latency?
by groxter on 2011-06-11 12:15:17

(2nd edit, but hey as long as no one else responded yet I may as well correct my original post and pretend I didn't answer it in a hurry the first time.)

The TCDDK software and hardware guide with specifications are downloadable ( The codec ( analog filter group delay is 32 sample periods, then there's transmission time to and from the codec, 1 sample period each way, and buffered i/o copy time. Let's call it 36 sample periods total latency (if I'm way off someone should say so.)  At 39.0625 kHz this is 0.92 ms, so it's right on the edge for your application.

Of course, you can increase the sample rate by modifying the clock division settings from the Line 6 template code, for lower latency but fewer compute cycles per sample.

Extremely low latency for realtime playing is one reason I prefer the ToneCore platform to writing DAW plugins, where you can easily encounter OS-incurred latencies of hundreds of samples.

Re: Input-output audio latency?
by audiofanatic on 2011-06-12 03:11:01

Thanks for your updated response. (Actually, I had noticed about the original post that didn't seem to include all 32 samples of group delay.) For me, it was very useful to know which codec it is.

One question to consider is whether the software programming environment allows sound to be processed in blocks of 1 sample. For instance, it might have some minimum blocksize, either for software efficiency reasons, or due to requirements for communication over the serial bus, which of course also adds additional latency.

The information above may not be current, and you should direct questions to the current forum or review the manual.