4 ms·

Why do you need the lowest latency possible to operate a DAW?

prmoustache 2y ago
It is important if you are driving hardware synths and samplers through midi. Otherwise latency is not really a problem.
- deleted 2y ago
  [deleted]
- wdfx 2y ago
  > Otherwise latency is not really a problem This is absolutely not true. Latency in audio systems is important in almost every aspect of using a DAW
- jcelerier 2y ago
  it's pretty hard for me as a musician to record guitar tracks in sync if I'm not getting the lowest latency through my computer (since I try to use e.g. software amp sims & pedals, etc.). Past ~8ms I feel it when I play, past 15ms I can hear the less accurate playing in the recorded tracks.
  - Aldipower 2y ago
    Use direct monitoring of your interface or an mixer in front of your interface. Then you can get away with 0ms (an analog mixer) or 2-4ms (interface direct monitor). Your DAW will latency compensate the "record head". It is just a matter of signal routing. It took me 20 years to get to this simple solution. :-D
    - jcelerier 2y ago
      > Use direct monitoring of your interface but that doesn't allow me to have my (software) effects chain when I record ? for a guitar solo where i'm going to play with say, delays and whammys that's a no-go
TehCorwiz 2y ago
There are three places to accumulate latency, the input from the instrument to the computer, the processing of filters, the output to the monitor (headphones/speakers)/recording. Sometimes you can get away with 2 or 3 ms of latency, but anything over 5 ms is super frustrating. Remember, you're fighting the latency between plucking a string or hitting a key and when the computer acknowledging the data and sending it back out to the monitor which you're using as your guide. Best case you go and "massage" the new track to line up with the existing tracks, worst case it sounds like an out of sync high school marching band. EDIT: The concerns here are primarily with input latency. Between plucking a string and heading it in your monitor it has to go through: your input hardware, the USB interface, the OS, the browser (which doesn't have explicit low latency capabilities), and JS. Most platforms support ASIO which is a low-level driver for reading audio data from devices. About as close to reading the ADCs yourself. Without a low-latency driver working with the OS there's so much latency overhead it's audible.
- Mashimo 2y ago
  Thanks. AFAIK ASIO is windows only.
  - TehCorwiz 2y ago
    I think you're right. I remember trying to use it on Ubuntu about a decade ago, but it seems that was through some wrapper.
    - gordy_gordstein 2y ago
      JACK Audio Connection Kit is generally the way to go on Linux. Arch wiki has a page on "Professional Audio" with more detail for the curious.
    - PaulDavisThe1st 2y ago
      There has never been ASIO on Linux. ASIO is a 3rd party (Steinberg) driver API exclusively for Windows.
wdfx 2y ago
Because when you want to record your instrument along with whatever else is in your project, timing is critical and everything needs to line up. You cannot be performing to audio that you are hearing with any delay, especially if the monitoring of the live audio is also being routed through software. At a certain point of latency it introduced delays and badly affects how you perform. In some circumstances it makes performance actually impossible. There are ways around this, namely if the software knows exactly what the input and output latency is then the playback and recording can be compensated. For live monitoring though you really need that done in the audio hardware itself in hard real time.
tigeba 2y ago
Music is very latency sensitive. If you are recording any source you generally want to have overall latency < 5ms. Input and monitoring latency is usually either handled by using fancy DSP systems or a "hack" where input audio bypasses any internal processing and gets routed directly back for monitoring.
BizarroLand 2y ago
I think the other people got a little too technical. The reasons are things like, if you want to play in time with a previously recorded track, or if you are using digital effects and need to be able to hear their effect on your instrument as you play it. Both of these are less of an issue today than they were 15 years ago where a USB 2.0 audio interface added significant delay into audio and made it harder to get what you wanted out of the system.

It is important if you are driving hardware synths and samplers through midi. Otherwise latency is not really a problem.

deleted 2y ago
[deleted]
wdfx 2y ago
> Otherwise latency is not really a problem This is absolutely not true. Latency in audio systems is important in almost every aspect of using a DAW
jcelerier 2y ago
it's pretty hard for me as a musician to record guitar tracks in sync if I'm not getting the lowest latency through my computer (since I try to use e.g. software amp sims & pedals, etc.). Past ~8ms I feel it when I play, past 15ms I can hear the less accurate playing in the recorded tracks.
- Aldipower 2y ago
  Use direct monitoring of your interface or an mixer in front of your interface. Then you can get away with 0ms (an analog mixer) or 2-4ms (interface direct monitor). Your DAW will latency compensate the "record head". It is just a matter of signal routing. It took me 20 years to get to this simple solution. :-D
  - jcelerier 2y ago
    > Use direct monitoring of your interface but that doesn't allow me to have my (software) effects chain when I record ? for a guitar solo where i'm going to play with say, delays and whammys that's a no-go

TehCorwiz 2y ago

There are three places to accumulate latency, the input from the instrument to the computer, the processing of filters, the output to the monitor (headphones/speakers)/recording. Sometimes you can get away with 2 or 3 ms of latency, but anything over 5 ms is super frustrating. Remember, you're fighting the latency between plucking a string or hitting a key and when the computer acknowledging the data and sending it back out to the monitor which you're using as your guide. Best case you go and "massage" the new track to line up with the existing tracks, worst case it sounds like an out of sync high school marching band. EDIT: The concerns here are primarily with input latency. Between plucking a string and heading it in your monitor it has to go through: your input hardware, the USB interface, the OS, the browser (which doesn't have explicit low latency capabilities), and JS. Most platforms support ASIO which is a low-level driver for reading audio data from devices. About as close to reading the ADCs yourself. Without a low-latency driver working with the OS there's so much latency overhead it's audible.

Mashimo 2y ago
Thanks. AFAIK ASIO is windows only.
- TehCorwiz 2y ago
  I think you're right. I remember trying to use it on Ubuntu about a decade ago, but it seems that was through some wrapper.
  - gordy_gordstein 2y ago
    JACK Audio Connection Kit is generally the way to go on Linux. Arch wiki has a page on "Professional Audio" with more detail for the curious.
  - PaulDavisThe1st 2y ago
    There has never been ASIO on Linux. ASIO is a 3rd party (Steinberg) driver API exclusively for Windows.

wdfx 2y ago

Because when you want to record your instrument along with whatever else is in your project, timing is critical and everything needs to line up. You cannot be performing to audio that you are hearing with any delay, especially if the monitoring of the live audio is also being routed through software. At a certain point of latency it introduced delays and badly affects how you perform. In some circumstances it makes performance actually impossible. There are ways around this, namely if the software knows exactly what the input and output latency is then the playback and recording can be compensated. For live monitoring though you really need that done in the audio hardware itself in hard real time.

tigeba 2y ago

Music is very latency sensitive. If you are recording any source you generally want to have overall latency < 5ms. Input and monitoring latency is usually either handled by using fancy DSP systems or a "hack" where input audio bypasses any internal processing and gets routed directly back for monitoring.

BizarroLand 2y ago

I think the other people got a little too technical. The reasons are things like, if you want to play in time with a previously recorded track, or if you are using digital effects and need to be able to hear their effect on your instrument as you play it. Both of these are less of an issue today than they were 15 years ago where a USB 2.0 audio interface added significant delay into audio and made it harder to get what you wanted out of the system.