Latency?
- Thread starter twonky
- Start date
Stealthtech
LOGIC ABUSER
Latency is normally an issue with sound cards and interfaces. It's a delay in milliseconds of the sound going thru the gear to the output. Can be corrected by low latency gear and adjusting buffer settings.
twonky said:
It takes time (even if its 1/1000 of a second) for sound to come through an analog input, be converted to digital information and be converted back to analog audio so you can hear it - that is what latency is all about.
There is no way to completely eliminate it - more expensive soundcards just make the process go quicker.
Slackmaster2K
Gone
Latency is only an issue when:
Mixing - it is the time it takes for you to hear the result of a change (e.g. you slide a fader). For mixing, latency values in the <80ms range are decent, and any WDM / ASIO soundcard+software combo in just about any computer can do this.
Live Input Processing - when applying effects in realtime to incoming audio. Say you want to add a reverb plugin to your guitar track and monitor it WHILE you're actually playing the guitar track. In this case you want latency as low as possible, but as a rule anything < 10ms is ok. Most decent ASIO/WDM soundcards in semi-fast systems can make this happen to a certain extent...you might not get latency that low if you're running a huge project.
Virtual Instruments - latency in this case is basically the same as with live input processing. It's the amount of time it takes for you to hear a sound after pressing a key on your keyboard (etc). Again, latency as low as possible is good, with <10ms being the rule of thumb. Good soundcard, ASIO/WDM, etc etc.
Latency is not normally an issue when just doing "normal" recording. That's because *most* soundcards support a feature called "Zero Latency Monitoring", which means that the card's outputs are routed to its inputs BEFORE the audio signal even hits the converters. The downside of this is that you don't really hear the exact signal as it's recorded (e.g. you don't hear what the converters do to the sound until you play it back), but the upside is that you can overdub using high latency settings without having to compensate.
Slackmaster 2000
Mixing - it is the time it takes for you to hear the result of a change (e.g. you slide a fader). For mixing, latency values in the <80ms range are decent, and any WDM / ASIO soundcard+software combo in just about any computer can do this.
Live Input Processing - when applying effects in realtime to incoming audio. Say you want to add a reverb plugin to your guitar track and monitor it WHILE you're actually playing the guitar track. In this case you want latency as low as possible, but as a rule anything < 10ms is ok. Most decent ASIO/WDM soundcards in semi-fast systems can make this happen to a certain extent...you might not get latency that low if you're running a huge project.
Virtual Instruments - latency in this case is basically the same as with live input processing. It's the amount of time it takes for you to hear a sound after pressing a key on your keyboard (etc). Again, latency as low as possible is good, with <10ms being the rule of thumb. Good soundcard, ASIO/WDM, etc etc.
Latency is not normally an issue when just doing "normal" recording. That's because *most* soundcards support a feature called "Zero Latency Monitoring", which means that the card's outputs are routed to its inputs BEFORE the audio signal even hits the converters. The downside of this is that you don't really hear the exact signal as it's recorded (e.g. you don't hear what the converters do to the sound until you play it back), but the upside is that you can overdub using high latency settings without having to compensate.
Slackmaster 2000
Slack,
I'm glad you mentioned that last part, because for the longest time when I first started I didn't realize that I could use the monitoring until when day I was tinkering with my set up. Before I used monitoring I used the regular output method which sucked to deal with the latency, but since using the monitoring method, latency is basically a non-issue for me.
And, because latency is not an issue for me anymore, I can put the buffer at a higher range which helps prevent dropouts while working
micro.
I'm glad you mentioned that last part, because for the longest time when I first started I didn't realize that I could use the monitoring until when day I was tinkering with my set up. Before I used monitoring I used the regular output method which sucked to deal with the latency, but since using the monitoring method, latency is basically a non-issue for me.
And, because latency is not an issue for me anymore, I can put the buffer at a higher range which helps prevent dropouts while working
micro.
DeadPoet
carpe diem
Slack: I don't know about this 10ms thing... If I play in direct into the computer (bass that is) I swear I hear and (more important) feel a delay when I'm at 5ms!
I once set the sampling rate on my Cport to 96k (=1ms) and the lag was gone.
then again, I had this teacher in conservatory who didn't want to play on Roland keyboards, because they lagged when he was playing 128th notes...
Heriwg
I once set the sampling rate on my Cport to 96k (=1ms) and the lag was gone.
then again, I had this teacher in conservatory who didn't want to play on Roland keyboards, because they lagged when he was playing 128th notes...
Heriwg
Slackmaster2K
Gone
Well sure! 10ms is the figure typically given that represents the amount of time required for the human ear to hear two seperate sounds.
That doesn't mean that you won't sense a 10ms delay when playing your keyboard. I toss out that figure because it's usually at the very least usable to most people. If you're going to use a DAW for this kind of work, you have to do what you have to do!
Slackmaster 2000
That doesn't mean that you won't sense a 10ms delay when playing your keyboard. I toss out that figure because it's usually at the very least usable to most people. If you're going to use a DAW for this kind of work, you have to do what you have to do!
Slackmaster 2000