• SONAR
  • Another introduction! (p.3)
2013/09/02 08:46:50
jonboyuk
Good point - I'm not sure. I'll check and report back with my findings.
2013/09/02 09:33:26
doncolga
jonboyuk
I just want to say - the quad capture has arrive and it's AWESOME!!!!!!!! It works perfectly on both of my devices under USB 2.0 & 3.0!
 
One thing I don't get is something with the latency. Any insight would be great.
 
- At a sample rate of 48kHz, I get a return trip of 5.6ms
- At a sample rate of 96kHz, I get a return trip of 4.2ms
- At a sample rate of 192kHz, I get a return trip of 3.5ms!
 
No hiss or pops or crackles (as long as I have 'Reduce CPU load' ticked).
 
I prefer 96 but I get a better latency when it's higher. Why would this be? Logic says it should be the other way around!


That does seem backwards, but my RME Multiface does the same this as I just checked...that's very cool.  I'm not observed that before.
2013/09/02 10:03:15
ston
Down the cabling itself it's going to be nigh-on light speed, so the difference in latency isn't occurring there. 
 
Ah, could it be less interpolation calculations for higher sample rates hence lower latency?
2013/09/02 10:10:54
wizard71
Latency is measured in samples. The closer the samples are the less latency there is.

Bibs
2013/09/02 10:16:06
ston
wizard71
Latency is measured in samples. The closer the samples are the less latency there is.

Bibs

That's not right.  You can measure latency in any unit of time.  Latency is processing overhead, the delay caused by sampling, writing to buffers, reading from buffers and signal reconstruction.
 
Look at it this way, you generate a signal so short there is only one sample taken.  There is only one sample so there is no 'closer the samples are' to consider.  The latency is the time taken for the signal to be sampled then reconstructed and returned.
2013/09/02 10:24:23
wizard71
No it probably isn't right, I googled it :-)
2013/09/02 10:44:39
2:43AM
Welcome to the forum!
 
As for a DAW, there will be a learning curve associated with a change, e.g. if you decide to go with Sonar X2 (or X3 when it comes out sometime in the near future...at least that's what we're all suspecting). So if you're used to Ableton, and want to keep making music in a method you know how, then I'd say stick with Ableton. I demoed Ableton (the CD that came with my interface), and I disliked it something fierce! The GUI is for twerking aliens!
 
Cubase would be great to demo if Steinberg wouldn't require the purchase of an USB-eLicenser just to do so. And you can't demo the Elements version (which does not require the USB-eLicenser). Dumb!
2013/09/02 11:17:36
Jim Roseberry
jonboyuk
I just want to say - the quad capture has arrive and it's AWESOME!!!!!!!! It works perfectly on both of my devices under USB 2.0 & 3.0!
 
One thing I don't get is something with the latency. Any insight would be great.
 
- At a sample rate of 48kHz, I get a return trip of 5.6ms
- At a sample rate of 96kHz, I get a return trip of 4.2ms
- At a sample rate of 192kHz, I get a return trip of 3.5ms!
 
No hiss or pops or crackles (as long as I have 'Reduce CPU load' ticked).
 
I prefer 96 but I get a better latency when it's higher. Why would this be? Logic says it should be the other way around!




Sample-rate is the number of samples per second:
  • 48k is 48,000 samples per second
  • 96k is 96,000 samples per second
  • 192k is 192,000 samples per second
If the sample-rate in Sonar is set to 96k, it's playing back 96,000 samples per second.
 
 
There are two sources of latency with a host-based DAW:
  • Your audio interface
  • Any latent plugins
 
Let's say your audio interface is set to a 64-sample ASIO buffer size.  That buffer size is static (doesn't change).
The higher the sample-rate (more samples played per second), that 64-samples will have less latency.
You can calculate the latency of the ASIO buffer size (at any sample-rate) using algebra.
 
ie:  Using a 64-sample ASIO buffer size at 48k
64 (ASIO buffer size) divided by 48,000 (sample rate) = x (latency in ms) divided by 1000 (there are 1000ms in a second)
solve for x
The latency of a 64-sample ASIO buffer size at 48k is 1.33ms
 
Let's double the sample rate to 96k and do the same equation.
64 (ASIO buffer size) divided by 96,000 (sample rate) = x (latency in ms) divided by 1000 (there are 1000ms in a second)
solve for x
The latency of a 64-sample ASIO buffer size at 96k is 0.67ms
 
 
Round-trip latency is the sum of the following:
  • ASIO input buffer
  • ASIO output buffer
  • Latency from the A/D converters
  • Latency from the D/A converters
  • The audio interface driver's hidden safety buffer (this is the x-factor with audio interfaces)
 
Using the example of the 64-sample ASIO buffer size at 48k you get the following round-trip latency:
ASIO input buffer 1.3ms
ASIO output buffer 1.3ms
Latency from the A/D and D/A converters (very roughly 2ms)
The driver's hidden safety buffer (varies from very small to huge depending on the audio interface's driver)
The end user typically has no control over the hidden safety buffer (so there's no real way to mitigate its latency).
The best audio interfaces offer sub 5ms total round-trip latency at a 64-sample ASIO buffer size/48k
 
 
 
2013/09/02 11:38:41
doncolga
Awesome post Jim.  In the big picture, is the lower latency impacted by the machine working harder @ 96K?
2013/09/02 11:46:54
jonboyuk
Wow Jim, I think I'll have to read that a few times to understand it....but I do get the gist. 
 
Therefore in each case, 5.6/4.2/3.6 are actually all very respectable latencies and I should be over the moon? (I am incidentally)!
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