Record and mix your own music? Get your copy of my free guide "7 Recording Mistakes That Ruin Mixes" : www.dadrockandguitars.com/7-recording-mistakes-that-ruin-mixes
Awesome Video, thank you so much! I have always had this problem with my Midi USB Keyboard , once again thank you for posting the solution fantastic video!
Way Cool! At first look you seemed to be a nice guy you'd see at a hardware store or some mom and pop situation but it only took me a moment to realize that you were articulate, concise and knowledgeable. I have S1 Pro6 so I knew about the block size already. I didn't know about the rest! Thank you. I have direct monitoring on my Focusrite interface which is supposed to help with this by sending the headphone signal directly back to you as it enters the interface. This can sometimes still have glitches if the latency from the DAW is too high. Anyway: you have my attention and I like your uncluttered delivery, so I'll become a follower. JP
Thanks! How to fix the problem that when Z for instruments is enabled it sounds different or even less loud while playing. Track armed. Is this caused by the low latency mode?
Really good question! Without more info, it's a little hard for me to know why that's happening for you. However, if you are using multiple plugins on that channel, then it may be that some of the plugins are being disabled when Low Latency Monitoring is turned on. For a plugin to be active and heard during low latency monitoring, it must have 3 milliseconds or less of latency. So if there are plugins that have more than 3ms of latency, when they turn off, the sound will change because they are no longer affecting the sound. And if there is gain/volume being added in the plugin, then the sound would drop in volume as well. That's my best guess with the given info. Let me know if that helps or if you're still having issues.
Hello, I have a question regarding the settings that attempt to resolve latency in Studio One. The LOW LATENCY INSTRUMENT MONITORING option disables (turns off) the WAVES, CLA NX plugin. Any ideas why this happens and/or how to fix it? (Sorry for my bad English, I used Google Translate)
That's a great question! If the plugin causes more than 3 milliseconds of latency, then Low Latency Monitoring will turn the plugin off and you won't hear the effect. It looks like CLA has ~442 milliseconds of latency. I found this helpful chart from Waves that gives the latency for all of their plugins: www.waves.com/support/tech-specs/plugin-latency I have CLA NX as well. It would be nice to monitor through CLA NX, but it looks like it isn't possible with Low Latency Monitoring turned on. From Presonus: s1manual.presonus.com/en/#Setup_Topics/Set_Up_Your_Audio_Device.htm#Audio Plug-in Use with Native Low-Latency Monitoring When monitoring an audio input or virtual instrument through the Native Low-Latency Monitoring system, any inserted FX on the corresponding Channel continue to function and can be heard in real time, provided that they add 3 ms or less of latency. Plug-ins that meet this latency requirement show a green power button in the Console (rather than blue or gray). Any inserted plug-ins that introduce more than 3 ms of latency are not audible in the monitoring path while a Channel is armed for monitoring or recording under Native Low-Latency Monitoring. They begin functioning again when recording/monitoring mode is disengaged. I hope that helps! Let me know if you have any other questions. By the way, Google translate worked really well. Your English was better than most English speaking commenters. Ha ha!
I haven't used Reaper, but I found this video. Skip to 1:40 and that should show you what you need. ruclips.net/video/303eDz-8SW0/видео.htmlsi=_JC8g_dsnOadKjTf
Awesome video..do you think Macbook handles sound or latecy better? I have an Asus gaming laptop with decent specs with an interface and still have annoying level of latency 😅
Great question! I'm not sure if Mac is "better" but it does pretty much just handle it on its own. Windows usually requires a driver, but I'm pretty sure Windows is capable of handling the latency well. Are you using an audio interface? If so, have you installed the latest drivers/routing software for it? Does your recording software (I assume probably Studio One since you watched this video) show options for you to choose low latency? 64 samples, 128 samples, etc.?
I have never used Bandlab, but it looks like it doesn't really have a way to adjust the buffer size for improving latency during recording. I did find this Help Article that talks about latency a bit: help.bandlab.com/hc/en-us/articles/115002959414-How-do-I-troubleshoot-latency Hope that helps!
Good video; I’m using ezdrummer 3 standalone with an Alesis nitro mesh; but still a little latency specially when playing the kick. What should I do if I’m planning to go through the PA live? Do I need an interface? Or should I just go from the audio output of the computer to the board? Your answer would be a great help.
Regarding the latency, have you adjusted the Audio Buffer Size setting under Settings -> Audio/MIDI Setup? I would recommend using an audio interface with balanced outputs. Those outputs could go straight to the board. The audio interface would also be able to handle the MIDI connection as well. You might be able to get away with using the headphone output. You may have to play with the volume on your laptop since they headphone out is an amplified signal. You'd want a 3.5mm TRS Male to Dual 1/4-inch TS Male cable. That would be an unbalanced signal and if running much length at all (more than 20ft), could have noise issues. You'd want to use a couple of DI boxes to convert it to a balanced signal over XLR cables. Hopefully that helps. Let me know if you have additional questions.
Hey man not sure if you will see this but here goes. Current situation - Followed all instructions, using a Roland design Midi to USB cord direct to PC. Loaded up ASIO4ALL which has given me good buffer and frequency above the standard drivers first listed, for my TD 6V kit I am at 64 and 4800. There was also an internal switch on the module (recommended in forums to turn off) did so and noticed a small improvement in latency. Drums are triggering nicely, but occasionally I'll strike the snare or a tom and get a little bit of static also ASIO4ALL is good (has increased available buffer and allowed my to dial in). Would you have any suggestions or from community? so close but the occasional static and mix issues are frustrating - cheers!
I haven’t had that exact issue However, when I’m playing guitar direct and using an amp sim, if I have the buffer size too low and the computer is doing other stuff that requires a decent amount of CPU resources (playing a recording session back with lots of plug-ins for example), then it will produce static every now and then because the computer can’t keep up with all of the processing. In that case, I increase the buffer size a little and that usually fixes it. A buffer size of 64 is nice and low. If you raise it and the static goes away, then you’ll just have to find that balance where latency is as good as it can be and your computer can keep up. Let me know if that doesn’t work, but that’s my initial guess.
When I am using EZdrummer 3 standalon I try to decrease the buffer size to 256. But in EZdrummer. But I notice that if I open the Asio4All panel it says other buffer size, like 512. What is the buffer size that counts ? Or do I have to lower both ?
I'm not on Windows (Mac doesn't used the Asio4II driver) so I'm not 100% sure, but I think that you should only need to adjust that in EZdrummer. If you make the buffer size really high in EZD, does the latency get bad? And then if you set EZD to a smaller buffer size, does the latency get a lot better? If so, then I'd assume that EZD is handling the buffer size. Sorry I can't give you an absolute answer, but that's how I would test it. Hope that helps.
Hey Allan! Do you have a Presonus interface? If so, my understanding is that the Z will be blue because it's using the hardware. I don't have a Presonus Interface, therefore it's using my computer to do and is green in that case. From Presonus website: "When Native Low-Latency Monitoring is enabled, the "Z" button turns green. When Hardware Low-Latency Monitoring is enabled, the "Z" button turns blue." There are some links to Presonus in the description that go into a lot more detail if you want to check those out. Let me know if you have any more questions. Happy to help where I can!
@@DadRockAndGuitars thanks. No I have a RME interface. I found if the drop out protection is too low relative to the buffer size you will not get the green Z. Its like the drop out protection set to high sets a high buffer for all channels playing back. You can then set the device buffer size low and when you click the Z and it goes green you get high buffer on the play back tracks and low buffer on the track thats record armed. I didnt fully get it earlier. With all my messing around i was getting the blue Z but when i click on it, it would mute the armed track i guess cause as you said it wants to use a Presonus interface. Thanks for your reply, i think i have my issues sorted out
Record and mix your own music? Get your copy of my free guide "7 Recording Mistakes That Ruin Mixes" : www.dadrockandguitars.com/7-recording-mistakes-that-ruin-mixes
Finally I found a video that save me... thanks a lot!!!!
That's great! So glad the video was able to help you out 👍
happily playing my NUX220 with only 8 ms latency. Thankyou very much from Chile!
That's great! Glad you were able to get the latency so low! Thanks for watching 👍
This is just what I needed! God bless u :)
Great! Thanks for watching and letting me know it was helpful 👍
Awesome Video, thank you so much! I have always had this problem with my Midi USB Keyboard , once again thank you for posting the solution fantastic video!
Awesome! Glad you were able to get it working! Thanks for watching and appreciate you letting me know 👍
Way Cool! At first look you seemed to be a nice guy you'd see at a hardware store or some mom and pop situation but it only took me a moment to realize that you were articulate, concise and knowledgeable. I have S1 Pro6 so I knew about the block size already. I didn't know about the rest! Thank you. I have direct monitoring on my Focusrite interface which is supposed to help with this by sending the headphone signal directly back to you as it enters the interface. This can sometimes still have glitches if the latency from the DAW is too high. Anyway: you have my attention and I like your uncluttered delivery, so I'll become a follower. JP
Thanks JP! Glad the video was helpful and thanks for subscribing!
Awesome! Just what I needed. Thanks.
Glad it was helpful! Thanks for watching and letting me know 👍
Thank you brother just what I needed!
That's great! I'm glad it was helpful for you! Thanks for watching and letting me know 👍
Thanks that was a great help buddy...😉
Awesome! Glad it helped you out! Thanks for watching and letting me know 👍
Thanks!
How to fix the problem that when Z for instruments is enabled it sounds different or even less loud while playing. Track armed.
Is this caused by the low latency mode?
Really good question! Without more info, it's a little hard for me to know why that's happening for you. However, if you are using multiple plugins on that channel, then it may be that some of the plugins are being disabled when Low Latency Monitoring is turned on.
For a plugin to be active and heard during low latency monitoring, it must have 3 milliseconds or less of latency. So if there are plugins that have more than 3ms of latency, when they turn off, the sound will change because they are no longer affecting the sound. And if there is gain/volume being added in the plugin, then the sound would drop in volume as well.
That's my best guess with the given info. Let me know if that helps or if you're still having issues.
Amesome video and awesome info! 🙌 🔥
Thanks for watching 👍 Glad it was helpful!
Hello, I have a question regarding the settings that attempt to resolve latency in Studio One.
The LOW LATENCY INSTRUMENT MONITORING option disables (turns off) the WAVES, CLA NX plugin.
Any ideas why this happens and/or how to fix it?
(Sorry for my bad English, I used Google Translate)
That's a great question! If the plugin causes more than 3 milliseconds of latency, then Low Latency Monitoring will turn the plugin off and you won't hear the effect. It looks like CLA has ~442 milliseconds of latency. I found this helpful chart from Waves that gives the latency for all of their plugins: www.waves.com/support/tech-specs/plugin-latency
I have CLA NX as well. It would be nice to monitor through CLA NX, but it looks like it isn't possible with Low Latency Monitoring turned on.
From Presonus: s1manual.presonus.com/en/#Setup_Topics/Set_Up_Your_Audio_Device.htm#Audio
Plug-in Use with Native Low-Latency Monitoring
When monitoring an audio input or virtual instrument through the Native Low-Latency Monitoring system, any inserted FX on the corresponding Channel continue to function and can be heard in real time, provided that they add 3 ms or less of latency. Plug-ins that meet this latency requirement show a green power button in the Console (rather than blue or gray). Any inserted plug-ins that introduce more than 3 ms of latency are not audible in the monitoring path while a Channel is armed for monitoring or recording under Native Low-Latency Monitoring. They begin functioning again when recording/monitoring mode is disengaged.
I hope that helps! Let me know if you have any other questions.
By the way, Google translate worked really well. Your English was better than most English speaking commenters. Ha ha!
@@DadRockAndGuitars Thank you very much for your response, I hope this is the solution.
Hi, could you please help me find this option on reaper?
I haven't used Reaper, but I found this video. Skip to 1:40 and that should show you what you need.
ruclips.net/video/303eDz-8SW0/видео.htmlsi=_JC8g_dsnOadKjTf
@@DadRockAndGuitars thank you so much! It worked. 😄
@@rhinestoneeyed_ Awesome! Glad to help 👍
My Steven Slate drums cut off after a few hits. Why? my DAW is Waveform 12. Alesis Nitro kit
A little hard to know exactly why without more info. What’s your buffer size? If you increase it, then you’ll have latency, but does it still cut out?
Thankyou, I can solve it
That’s great 👍
Awesome video..do you think Macbook handles sound or latecy better?
I have an Asus gaming laptop with decent specs with an interface and still have annoying level of latency 😅
Great question! I'm not sure if Mac is "better" but it does pretty much just handle it on its own. Windows usually requires a driver, but I'm pretty sure Windows is capable of handling the latency well.
Are you using an audio interface? If so, have you installed the latest drivers/routing software for it?
Does your recording software (I assume probably Studio One since you watched this video) show options for you to choose low latency? 64 samples, 128 samples, etc.?
can i do this for bandlab? if so how do i do that?
I have never used Bandlab, but it looks like it doesn't really have a way to adjust the buffer size for improving latency during recording. I did find this Help Article that talks about latency a bit:
help.bandlab.com/hc/en-us/articles/115002959414-How-do-I-troubleshoot-latency
Hope that helps!
Great video thanks
Thanks for watching!
Good video; I’m using ezdrummer 3 standalone with an Alesis nitro mesh; but still a little latency specially when playing the kick.
What should I do if I’m planning to go through the PA live? Do I need an interface? Or should I just go from the audio output of the computer to the board? Your answer would be a great help.
Regarding the latency, have you adjusted the Audio Buffer Size setting under Settings -> Audio/MIDI Setup?
I would recommend using an audio interface with balanced outputs. Those outputs could go straight to the board. The audio interface would also be able to handle the MIDI connection as well.
You might be able to get away with using the headphone output. You may have to play with the volume on your laptop since they headphone out is an amplified signal. You'd want a 3.5mm TRS Male to Dual 1/4-inch TS Male cable. That would be an unbalanced signal and if running much length at all (more than 20ft), could have noise issues. You'd want to use a couple of DI boxes to convert it to a balanced signal over XLR cables.
Hopefully that helps. Let me know if you have additional questions.
thank you so much!
Thanks for watching Andrew!
Hey man not sure if you will see this but here goes. Current situation - Followed all instructions, using a Roland design Midi to USB cord direct to PC. Loaded up ASIO4ALL which has given me good buffer and frequency above the standard drivers first listed, for my TD 6V kit I am at 64 and 4800. There was also an internal switch on the module (recommended in forums to turn off) did so and noticed a small improvement in latency. Drums are triggering nicely, but occasionally I'll strike the snare or a tom and get a little bit of static also ASIO4ALL is good (has increased available buffer and allowed my to dial in). Would you have any suggestions or from community? so close but the occasional static and mix issues are frustrating - cheers!
I haven’t had that exact issue However, when I’m playing guitar direct and using an amp sim, if I have the buffer size too low and the computer is doing other stuff that requires a decent amount of CPU resources (playing a recording session back with lots of plug-ins for example), then it will produce static every now and then because the computer can’t keep up with all of the processing.
In that case, I increase the buffer size a little and that usually fixes it. A buffer size of 64 is nice and low. If you raise it and the static goes away, then you’ll just have to find that balance where latency is as good as it can be and your computer can keep up.
Let me know if that doesn’t work, but that’s my initial guess.
You can here it happening when I play guitar here: ruclips.net/video/B2e9ocegZRg/видео.htmlsi=k0DAYgc1BWvsRlRg&t=938
When I am using EZdrummer 3 standalon I try to decrease the buffer size to 256. But in EZdrummer. But I notice that if I open the Asio4All panel it says other buffer size, like 512. What is the buffer size that counts ? Or do I have to lower both ?
I'm not on Windows (Mac doesn't used the Asio4II driver) so I'm not 100% sure, but I think that you should only need to adjust that in EZdrummer. If you make the buffer size really high in EZD, does the latency get bad? And then if you set EZD to a smaller buffer size, does the latency get a lot better? If so, then I'd assume that EZD is handling the buffer size.
Sorry I can't give you an absolute answer, but that's how I would test it. Hope that helps.
@@DadRockAndGuitars yes, you are right. Ez drummer handles the buffer size. Thanks
Thanks man. Have you seen the "Z" go blue? Instead of green... Any ideas what it means. I seem to not have the option where the Z goes green?
Hey Allan! Do you have a Presonus interface? If so, my understanding is that the Z will be blue because it's using the hardware. I don't have a Presonus Interface, therefore it's using my computer to do and is green in that case.
From Presonus website: "When Native Low-Latency Monitoring is enabled, the "Z" button turns green. When Hardware Low-Latency Monitoring is enabled, the "Z" button turns blue."
There are some links to Presonus in the description that go into a lot more detail if you want to check those out.
Let me know if you have any more questions. Happy to help where I can!
@@DadRockAndGuitars thanks. No I have a RME interface.
I found if the drop out protection is too low relative to the buffer size you will not get the green Z. Its like the drop out protection set to high sets a high buffer for all channels playing back. You can then set the device buffer size low and when you click the Z and it goes green you get high buffer on the play back tracks and low buffer on the track thats record armed. I didnt fully get it earlier.
With all my messing around i was getting the blue Z but when i click on it, it would mute the armed track i guess cause as you said it wants to use a Presonus interface.
Thanks for your reply, i think i have my issues sorted out