needle ninja Posted October 21, 2008 Share Posted October 21, 2008 Spent the last two days fooling around with settings in Sonar... I managed to get 5.8ms latency using AISO drivers on my new gear. I may have to increase it for complex work though What are other people getting around here with different equipment? Quote Link to comment Share on other sites More sharing options...
Malevol3nt Posted October 21, 2008 Share Posted October 21, 2008 12ms on shitty hardware that I've got right now. But I can't even use it atm. Gonna get a FF400 sometime early next year. As for other equipment, I've seen it go as low as 1.5ms, that would be sweet to have (i think that was on the fireface but I can't be sure). Quote Link to comment Share on other sites More sharing options...
Otto Matta Posted October 21, 2008 Share Posted October 21, 2008 6.5-ish, and I've never adjusted it. Cubase 4, M-Audio Delta 1010LT, AMD Athlon X2 4800+. I freeze tracks when they start to cause problems, which can be a pain, but too much latency is like nails on a chalkboard for me. Quote Link to comment Share on other sites More sharing options...
Veracohr Posted October 21, 2008 Share Posted October 21, 2008 My DAW doesn't give times, just the buffer choice. I use 128 samples when I'm writing/playing. So, assuming there's no hidden latency that I'm missing, (1/44100)*128 = 2.9ms. I can use 64 samples to a point, but only if I've only got one or two processor-light instruments going in Reason, or a hardware one. Not a chance if I'm using a Native Instruments softsynth. When I mix, I jack the buffer up to 1024. Quote Link to comment Share on other sites More sharing options...
qa2pir Posted October 21, 2008 Share Posted October 21, 2008 I use latency as a creative device, much like I use windsurfing as a means of relaxing. Quote Link to comment Share on other sites More sharing options...
needle ninja Posted October 27, 2008 Author Share Posted October 27, 2008 I'm up to 7.2ms with a buffer of 384 right now. Some features of Sonar need more buffer, like Roland V-Vocal. Quote Link to comment Share on other sites More sharing options...
Amygdala Posted October 27, 2008 Share Posted October 27, 2008 My DAW doesn't give times, just the buffer choice. I use 128 samples when I'm writing/playing. So, assuming there's no hidden latency that I'm missing, (1/44100)*128 = 2.9ms. I can use 64 samples to a point, but only if I've only got one or two processor-light instruments going in Reason, or a hardware one. Not a chance if I'm using a Native Instruments softsynth. When I mix, I jack the buffer up to 1024. Depending on your driver, and the operation mode of this driver, it might do some internal buffering. This is extremely common, so it is only in rare cases you can trust the calculation above. Also, buffering happens on both the input and output side, so play-through latency is probably at least twice. Also, you should really consider going to a higher samplerate Anywho - I probably have a latency of about 20-30 ms. I don't really care that much about it. It is definetly noticeable, but I very rarely record stuff realtime, so it's no biggie for me. On the other hand, I can run thirty-some soft instruments with loads and loads of effects. *That* I like! Also, at very high bufferrates, online-processing-only plugins sound very cool when processed offline... BTW, Native Instruments plugs rock soooo hard! - A Quote Link to comment Share on other sites More sharing options...
Veracohr Posted October 28, 2008 Share Posted October 28, 2008 Depending on your driver, and the operation mode of this driver, it might do some internal buffering. This is extremely common, so it is only in rare cases you can trust the calculation above. Also, buffering happens on both the input and output side, so play-through latency is probably at least twice. Also, you should really consider going to a higher samplerate Anywho - I probably have a latency of about 20-30 ms. I don't really care that much about it. It is definetly noticeable, but I very rarely record stuff realtime, so it's no biggie for me. On the other hand, I can run thirty-some soft instruments with loads and loads of effects. *That* I like! Also, at very high bufferrates, online-processing-only plugins sound very cool when processed offline... BTW, Native Instruments plugs rock soooo hard! - A I use Audio Units drivers. I don't pay much attention to that side of things; I just get things working and go from there. As far as sample rate, there's no need. If it's going to end up on a CD, there's no point in recording at a higher rate. Especially with synthesizer sounds, which don't typically extend as high as acoustic sounds. And yes, Native Instruments makes good synths. Too bad Massive crashes my computer so I can't use it. Absynth and FM8 put a strain on it, but at least I can use them. I have an elderly computer, and can't afford a new one (since I use Mac). If I really wanted, I could use my much newer laptop (Macbook Pro 2.2GHz), but I think that would annoy the hell out of me, plus that would mean I can't use my UAD plugins (without spending money on the UAD Xpander). Quote Link to comment Share on other sites More sharing options...
Amygdala Posted October 28, 2008 Share Posted October 28, 2008 I use Audio Units drivers. I don't pay much attention to that side of things; I just get things working and go from there. As far as sample rate, there's no need. If it's going to end up on a CD, there's no point in recording at a higher rate. Especially with synthesizer sounds, which don't typically extend as high as acoustic sounds. And yes, Native Instruments makes good synths. Too bad Massive crashes my computer so I can't use it. Absynth and FM8 put a strain on it, but at least I can use them. I have an elderly computer, and can't afford a new one (since I use Mac). If I really wanted, I could use my much newer laptop (Macbook Pro 2.2GHz), but I think that would annoy the hell out of me, plus that would mean I can't use my UAD plugins (without spending money on the UAD Xpander). Yay, a fellow mac user... Nice Sorry about sounding like a know-it-all, but when talking sample- and bit-rates, the process is more important than the product. During processing and internally in Audio Units, a higher samplerate is beneficial. Better resolution means less damage to the signal, and almost any kind of processing can be considered "damaging" for the signal. So, even if you end up with a signal in 44.1 kHz as an end product, you should still seek to keep as good a resolution as possible in the process. This is regardless of the type of input signal (natural, synth, ...). The most dramatic change though is bit-depth. 16 bit is a bad starting point, since Audio Units always work in 32-bit. You might as well enjoy the rest of the dynamic space I can see, if you are stressed for computational power, then higher samplerates are kind of a sore point, but switching to 24 (or even 32) bit should mean no extra computational cost, since AUs are 32 bit only - depending on your host... Logic? If you use Logic you have no excuse. Of course, don't know if you are using 16 bit sampling or not.... Yup, Absynth is my absolut favourite! Massive is good too, although I haven't really begin using it. Shame on me, but Absynth just has so endlessly many possibilities... - A Quote Link to comment Share on other sites More sharing options...
frozen dream Posted October 28, 2008 Share Posted October 28, 2008 below 1ms, all the time do i get a prize?? :posford: internal dsp processing is bliss! dual core helps as well tho' Quote Link to comment Share on other sites More sharing options...
Recommended Posts
Join the conversation
You can post now and register later. If you have an account, sign in now to post with your account.