Note: please do not get distracted and confused by the "Additional Delay Compensation" value that can be seen in my ReaInsert instance at around 9:40 min.! This is NOT the correct value for my system! I got aware of having picked a wrong scene from the several screenvideos I had made only after uploading the final video - sorry!
@Bristolpa you create folders by dragging other tracks onto the track you want to have as the folder track. A blue line will appear. Depending on where you hold the tracks that you are dragging above the designated folder track, the blue line will show indentation, indicating at what child track hierarchy the dragged tracks will end up if you let go the mouse button. Have a look here: forum.cockos.com/showthread.php?t=134577. As I already pointed out in another post, I'm using a special Imperial theme edition that has the "show indented child tracks" feature which is a significant advantage to the original Imperial theme. You should be able to find it by Google search. btw, I'm very happy to help you with any of your issues, however, you will get answers much quicker if you ask via Google ("Reaper" followed by what you'd like to know) or in the very knowledgable and helpful Reaper forum where I'm hanging around also. TH-cam comment area is not the ideal place for this.
what a good job, I liked it a lot, very well explained and by the way I liked the work with Bae, thank you very much for taking the time to do it, a hug
P.S. the reason it was off by 1 sample is you kept selecting the waveform 1 sample early, when it was still below the zero point. At 6:20 you went to check if you got the start right, you didn't, but you thought you did. Other than that, PERFECT!!
Thanks for putting this together. A couple questions: 1. When you imploded the two items into takes to compare them wouldn't that cause both items to be run through the EQ again since the insert wasn't removed? 2. How did you configure your track control panel to show the inserts in red? 3. Lastly, would changing the sample rate affect the process of getting an accurate latency? Is it best to use the highest sample rate supported by the interface first?
Thanks for your comment, Ian! 1. Correct. That's why I toggled the track fx of the ReaInsert track to bypass at 20:05 (track fx button and bypass button turning red) ;-) 2. Not sure which red inserts you are refering to. Are those the ones I mentioned under 1.? I'm using WT_Imperial theme incl. the folder track mod. 3. The latency test shown has to be repeated for every sample rate and also for every change of the buffer size you might be using. I did multiple tests all in a row while the loop-back cabling was still in place and keep all different results in a list as a reference to be able to look the corresponding values up whenever I need them. I'm not a fan of high sample rates (88.2k, 96k, etc.). I'm using 48k for various reasons. I never heard a aonic difference that whas significant enough when testing 96k. However, running my system at 96k cuts the no. of inputs and outputs of most of my digital gear in half which is a major inconvenience. 48k and 24 bit yields great results and keeps file sizes fairly small. If you are always producing content that will get released on CD, you may clock your interface/audio system at 44.1k to avoid the sample rate conversion.
@@kyrorocks By using the right latency compensation value for a given interface it will show "zero offset" with regard to the reference signal. This is not the same as being "zero latency", though ;-) But at least, your recordings will now land at the right (intended) spot which will make them sound more groovy due to being connected tighter to the other tracks.
Thanks, Jennifer! They are printed with a laser printer. I've created a LibreOffice Calc spread-sheet where the cells have the exact width of the patchbay channels and half the height of the space between the upper and lower jack connectors. For the Samson S-Patch, 4x 8 channels fit before there's a little gap, so I stack these 4x 8 channels in the spread-sheet and cut them out accordingly later. The labels are actually covering the normalling switches. Each of the labels is attached with two small glue pads (soft, white, adhesive compound that you can knead) in order to be easily removable temporarily in case I need to toggle a switch. PM me in the Reaper forum if you are interested in the ods file. During Covid lockdown, I replaced the 4 Samson patchbays by 2 Ghielmetti ones and re-cabled the entire control room. You can only do something like this during a pandemic, LOL. Here's a video showcasing the Ghielmettis and their new LibreOffice layout: th-cam.com/video/ve_ytqdrHwk/w-d-xo.html
thanks for this vid. can you tell me how much turning the output slider down using rea insert plugin is normal ? im trying to figure out if my 3rd gen 18i20 has enough headroom coming back in from my stam master bus compressor. mine was 1140 sampls
Thanks for your comment, Daniel! It actually depends on several factors. I usually don't lower ReaInsert's input or output faders at all. It depends on whether your interface can handle high output (non-clipping) volume without distorting the signal. Sone cheaper interfaces can introduce signal distortion even below 0 dB due to too little analog headroom. It also depends on whether the analog gear connected to the interface is balanced or unbalanced. In case of balanced gear (preferred) set it to a ref level of +4 dB, if possible (some gear is switchable for use with balanced or unbalanced conncections/levels). The analog level should then be fine and not clip the input of your analog gear. I keep ReaInsert's ins and outs at 0 dB (unity) level since I can then also print the processed return signal at unity level so that it's easy to use that signal during mixdown instead of the original one without having to change the mix a lot or at all. I wouldn't recommend using ReaInsert's output fader to "push" or "drive" the analog gear. You could clip your converter. Rather have them a unity gain and insert another analog device that provides a gain feature at the start of your outboard chain (in front of the device you actually wanna push at its input stage). Always push levels into analog gear using analog gain, not digital gain. Then, set output gain of the final analog device in the analog processing chain to a volume that matches the one you see on ReaInsert's hardware send meter. If you then bypass ReaInsert, you shouldn't hear a lot of level change but a change in sound (reflecting what the analog processing chain has done to your signal). Obviously, neither of ReaInsert 's level meters should ever clip!
hi, have u tried put multiple hardware insert on yo mixng project ? My reaper works fine with just on simple track with hardware insert, but if i slam to many on different track it got messed up real bad.(especially on complex folder layers project) is thrr anything I need to configure ? plz help lol
no, I never used more than one instance at a time. I always print the processed signal before proceeeding to use ReaInsert on another track. Is it absolutely necessary for you to use multiple instances of ReaInsert simultaneously?
@@SonicAxiom A) GREAT VIDEOS!! B) none of this is 'absolutely necessary'.... We're all building our ideal Hybrid Systems, and have tons of analog gear to interface. I can see running a channel or a bus through 1 or 2 hardware inserts, then printing the results immediately. I do that. But that's not the holy grail. I'm gonna break this out into a new comment....
I have a trouble. I have Audient iD4, And my click recorded track is earlier then click source)))) I know it sounds weird, but I can not understand what a problem. I have M4 Sequoia Mac
It's not uncommon that some setups produce recordings that come earlier than the signal that gets played back. Obviously, the audio interface can't look into the future but since the computer is constantly buffering a portion of the sound prior to playing it back (thus, it already sort of knows what the incoming signal will be), it can happen that the timing of your round-trip signal is off in the negative direction. You can still compensate in the same way shown in the video, however, you have to put the negative sign in front of your compensation value.
@@eduardoandrade6353 So yes, you should run through the entire test procedure I'm showing in the video to determin the actual specific latency of the UMC 22 and properly compensate for it. However, due to the fact that your interface only has two outputs, you have to first disconnect your speakers from the UMC 22's outputs for the time you are doing the test. Then, connect a cable from the UMC 22's right output directly to its instrument input 2. You can use a simple guitar (jack) cable. Turn off direct monitoring on the UMC 22 and set the output gain knob to 12 o'clock. Uncheck "use audio driver reported latency" in Reaper Preferences -> Audio -> Recording and make sure that all correction values on that page are set to 0. Hit play in Reaper to play back the glued click item and adjust the input level on the UMC 22 with input level knob 2. When the level is ok, create a new track underneath the click track. Set the track's input to input 2, mono. Make sure that the track's monitoring button is not engaged to avoid a feedback loop!!! Now, record the incoming click signal on the new track. After the recording, zoom into the waveform until you see the samples (dots) and verify if the original click's and the newly recorded signal's waveforms line up perfectly. If not, follow the directions given in the video until both waveforms do allign accurately. After having compensated for your interface's latency, recordings will be recorded at the correct spot in Reaper. When recording your e-drums, you might have to engage direct monitoring on the Behringer and avoid to activate the monitoring on the track you are recording on in Reaper to be able to hear yourself without distracting latency echo. Good Luck and have fun!
Thank you, Luc! The compensation needs to be active constantly. You will have to change the value (by doing other latency tests) if you decide to change your audio device's buffer size or sample rate! I made my own list of different latency compensation values for all buffer sizes and sample rate changes I probably need to use from time to time to be able to quickly look up the corresponding value. You should also always do a manual ping in ReaInsert once before using it in a given situation to account for even miniscule latency variations.
Hello! Thank you for the video. I seem to have an issue where the samples my sound card is off by fluctuates. It will give me a consistent value for a short period of time and then it will be off by -1 or 5. Has this happened to you? If not, what hardware are you using?
@@AleArzMusic while a "mono" (= unbalanced) cable may pick up hum easier than a balanced one, this fact is of little importance when doing the loopback test. A little bit of noise in the recorded click will not alter the result with regard to latency.
Hi, thanks for this post, it's great! Though I'm having an issue. I've followed your instructions and works really well with the click track. However, when I add in my outboard gear (in this case a reverb pedal), the latency changes slightly. In fact, it works better when setting up outputs directly on the track, rather than in realnsert. Wondering if I'm doing something wrong?
Thanks for your comment, James. I can't say if you are doing something wrong (too little information). Basically, a reliable interface should always yield the same latency regardless the method you use to route signals through it. However, it's important to measure and compare the correct signal path (to avoid comparing apples with oranges)! A reverb pedal might be a special case due to its nature. Introducing some amount of latency is part of its design. It's not a good idea to use such a device to determin the latency of a system or signal path. The added latency you are experiencing is probably caused by the pre-delay the pedal is supposed to add to the signal. If your basic latency measurement of the same signal path is correct, it' should also work correctly with the pedal inserted into it.
no, a particular latency compensation value is only depending on the sample rate and the buffer size. Before running the latency test, you may wanna heavily stress your DAW/system with lots of plugins and tracks to find out, what a safe buffer size is for your situation without getting any audio glitches. Once this is done, run the latency test to find out the exact compensation value for that safe buffer size. If you ever change buffer size or sample rate later on, you will have to re-run the test under the new condition and will find out another latency compensation value only valid for the new conditions.
Note: please do not get distracted and confused by the "Additional Delay Compensation" value that can be seen in my ReaInsert instance at around 9:40 min.! This is NOT the correct value for my system! I got aware of having picked a wrong scene from the several screenvideos I had made only after uploading the final video - sorry!
@Bristolpa yes, however, it's entirely in German except for my VST plugin page: www.audioworld.de/AudioWorld%20VST-Plugins_e.php
@Bristolpa you create folders by dragging other tracks onto the track you want to have as the folder track. A blue line will appear. Depending on where you hold the tracks that you are dragging above the designated folder track, the blue line will show indentation, indicating at what child track hierarchy the dragged tracks will end up if you let go the mouse button. Have a look here: forum.cockos.com/showthread.php?t=134577.
As I already pointed out in another post, I'm using a special Imperial theme edition that has the "show indented child tracks" feature which is a significant advantage to the original Imperial theme. You should be able to find it by Google search.
btw, I'm very happy to help you with any of your issues, however, you will get answers much quicker if you ask via Google ("Reaper" followed by what you'd like to know) or in the very knowledgable and helpful Reaper forum where I'm hanging around also. TH-cam comment area is not the ideal place for this.
The best video on TH-cam, about how really use output gear on Reaper. Thanks a lot, works very well for my equipments, and my DAW.
Thank you so much for your kind comment, Douglas! All the best!
what a good job, I liked it a lot, very well explained and by the way I liked the work with Bae, thank you very much for taking the time to do it, a hug
Thanks for your kind words! Much appreciated! Glad you liked it.
P.S. the reason it was off by 1 sample is you kept selecting the waveform 1 sample early, when it was still below the zero point. At 6:20 you went to check if you got the start right, you didn't, but you thought you did. Other than that, PERFECT!!
Thank you for this!
You're welcome! Thanks for watching!
Thanks for putting this together. A couple questions:
1. When you imploded the two items into takes to compare them wouldn't that cause both items to be run through the EQ again since the insert wasn't removed?
2. How did you configure your track control panel to show the inserts in red?
3. Lastly, would changing the sample rate affect the process of getting an accurate latency? Is it best to use the highest sample rate supported by the interface first?
Thanks for your comment, Ian!
1. Correct. That's why I toggled the track fx of the ReaInsert track to bypass at 20:05 (track fx button and bypass button turning red) ;-)
2. Not sure which red inserts you are refering to. Are those the ones I mentioned under 1.? I'm using WT_Imperial theme incl. the folder track mod.
3. The latency test shown has to be repeated for every sample rate and also for every change of the buffer size you might be using. I did multiple tests all in a row while the loop-back cabling was still in place and keep all different results in a list as a reference to be able to look the corresponding values up whenever I need them.
I'm not a fan of high sample rates (88.2k, 96k, etc.). I'm using 48k for various reasons. I never heard a aonic difference that whas significant enough when testing 96k. However, running my system at 96k cuts the no. of inputs and outputs of most of my digital gear in half which is a major inconvenience. 48k and 24 bit yields great results and keeps file sizes fairly small. If you are always producing content that will get released on CD, you may clock your interface/audio system at 44.1k to avoid the sample rate conversion.
Very good video. Thanks for sharing this!
thank you for watching, KYRO. Glad you found it interesting.
@@SonicAxiom I tried the latency test. Zero latency through my Roland Octa Capture! I knew it was low but that's great :)
@@kyrorocks By using the right latency compensation value for a given interface it will show "zero offset" with regard to the reference signal. This is not the same as being "zero latency", though ;-) But at least, your recordings will now land at the right (intended) spot which will make them sound more groovy due to being connected tighter to the other tracks.
Works like a charm. Thanks.
Very useful thank you. How do you print your patch bay labels? They look great!
Thanks, Jennifer! They are printed with a laser printer. I've created a LibreOffice Calc spread-sheet where the cells have the exact width of the patchbay channels and half the height of the space between the upper and lower jack connectors. For the Samson S-Patch, 4x 8 channels fit before there's a little gap, so I stack these 4x 8 channels in the spread-sheet and cut them out accordingly later. The labels are actually covering the normalling switches. Each of the labels is attached with two small glue pads (soft, white, adhesive compound that you can knead) in order to be easily removable temporarily in case I need to toggle a switch. PM me in the Reaper forum if you are interested in the ods file.
During Covid lockdown, I replaced the 4 Samson patchbays by 2 Ghielmetti ones and re-cabled the entire control room. You can only do something like this during a pandemic, LOL. Here's a video showcasing the Ghielmettis and their new LibreOffice layout: th-cam.com/video/ve_ytqdrHwk/w-d-xo.html
thanks for this vid. can you tell me how much turning the output slider down using rea insert plugin is normal ? im trying to figure out if my 3rd gen 18i20 has enough headroom coming back in from my stam master bus compressor. mine was 1140 sampls
Thanks for your comment, Daniel! It actually depends on several factors. I usually don't lower ReaInsert's input or output faders at all. It depends on whether your interface can handle high output (non-clipping) volume without distorting the signal. Sone cheaper interfaces can introduce signal distortion even below 0 dB due to too little analog headroom. It also depends on whether the analog gear connected to the interface is balanced or unbalanced. In case of balanced gear (preferred) set it to a ref level of +4 dB, if possible (some gear is switchable for use with balanced or unbalanced conncections/levels). The analog level should then be fine and not clip the input of your analog gear. I keep ReaInsert's ins and outs at 0 dB (unity) level since I can then also print the processed return signal at unity level so that it's easy to use that signal during mixdown instead of the original one without having to change the mix a lot or at all.
I wouldn't recommend using ReaInsert's output fader to "push" or "drive" the analog gear. You could clip your converter. Rather have them a unity gain and insert another analog device that provides a gain feature at the start of your outboard chain (in front of the device you actually wanna push at its input stage). Always push levels into analog gear using analog gain, not digital gain.
Then, set output gain of the final analog device in the analog processing chain to a volume that matches the one you see on ReaInsert's hardware send meter. If you then bypass ReaInsert, you shouldn't hear a lot of level change but a change in sound (reflecting what the analog processing chain has done to your signal). Obviously, neither of ReaInsert 's level meters should ever clip!
You Are THE BEST
Thanks, Marcin! Glad you found the video useful.
hi, have u tried put multiple hardware insert on yo mixng project ? My reaper works fine with just on simple track with hardware insert, but if i slam to many on different track it got messed up real bad.(especially on complex folder layers project) is thrr anything I need to configure ? plz help lol
no, I never used more than one instance at a time. I always print the processed signal before proceeeding to use ReaInsert on another track. Is it absolutely necessary for you to use multiple instances of ReaInsert simultaneously?
@@SonicAxiom A) GREAT VIDEOS!! B) none of this is 'absolutely necessary'....
We're all building our ideal Hybrid Systems, and have tons of analog gear to interface. I can see running a channel or a bus through 1 or 2 hardware inserts, then printing the results immediately. I do that. But that's not the holy grail.
I'm gonna break this out into a new comment....
Great info thanks
Thanks for your comment. Glad you found it useful!
I have a trouble. I have Audient iD4, And my click recorded track is earlier then click source)))) I know it sounds weird, but I can not understand what a problem. I have M4 Sequoia Mac
It's not uncommon that some setups produce recordings that come earlier than the signal that gets played back. Obviously, the audio interface can't look into the future but since the computer is constantly buffering a portion of the sound prior to playing it back (thus, it already sort of knows what the incoming signal will be), it can happen that the timing of your round-trip signal is off in the negative direction. You can still compensate in the same way shown in the video, however, you have to put the negative sign in front of your compensation value.
Hi, i use a e-drum connected to a interface through a USB port. Are this instructions appropriate to me?
Sorry for english!
Hi Eduardo, which interface are you using exactly? In principle, the instructions are valid for every audio interface.
@@SonicAxiom
Its a Behringer UMC 22
@@eduardoandrade6353 So yes, you should run through the entire test procedure I'm showing in the video to determin the actual specific latency of the UMC 22 and properly compensate for it. However, due to the fact that your interface only has two outputs, you have to first disconnect your speakers from the UMC 22's outputs for the time you are doing the test. Then, connect a cable from the UMC 22's right output directly to its instrument input 2. You can use a simple guitar (jack) cable. Turn off direct monitoring on the UMC 22 and set the output gain knob to 12 o'clock. Uncheck "use audio driver reported latency" in Reaper Preferences -> Audio -> Recording and make sure that all correction values on that page are set to 0. Hit play in Reaper to play back the glued click item and adjust the input level on the UMC 22 with input level knob 2. When the level is ok, create a new track underneath the click track. Set the track's input to input 2, mono. Make sure that the track's monitoring button is not engaged to avoid a feedback loop!!! Now, record the incoming click signal on the new track. After the recording, zoom into the waveform until you see the samples (dots) and verify if the original click's and the newly recorded signal's waveforms line up perfectly. If not, follow the directions given in the video until both waveforms do allign accurately.
After having compensated for your interface's latency, recordings will be recorded at the correct spot in Reaper. When recording your e-drums, you might have to engage direct monitoring on the Behringer and avoid to activate the monitoring on the track you are recording on in Reaper to be able to hear yourself without distracting latency echo.
Good Luck and have fun!
very good, thank you for this vidéo . question: the feast of the compensation of the delay made, we must leave all the time, or just for the reamping?
Thank you, Luc! The compensation needs to be active constantly. You will have to change the value (by doing other latency tests) if you decide to change your audio device's buffer size or sample rate! I made my own list of different latency compensation values for all buffer sizes and sample rate changes I probably need to use from time to time to be able to quickly look up the corresponding value. You should also always do a manual ping in ReaInsert once before using it in a given situation to account for even miniscule latency variations.
@@SonicAxiom OK thanks for your answer . but it must be active even if I record live guitar, bass, drums .. etc ... even if I do not reamping?
@@lucodin5205 yes, it makes sure that all your recordigs land on the right spot on the timeline in Reaper.
@@SonicAxiom thank you very much
Hello! Thank you for the video. I seem to have an issue where the samples my sound card is off by fluctuates. It will give me a consistent value for a short period of time and then it will be off by -1 or 5. Has this happened to you? If not, what hardware are you using?
Will definitely do this. Any particular jack cable to use for the re insertion?
no, wether you use XLR, jack or something else only depends on the type of connectors you have on your gear.
@@SonicAxiom thank you for answering me so quickly! I'm using a scarlett 2i2 2nd Gen, Reaper 6, cheap mono jack cable
@@AleArzMusic while a "mono" (= unbalanced) cable may pick up hum easier than a balanced one, this fact is of little importance when doing the loopback test. A little bit of noise in the recorded click will not alter the result with regard to latency.
@@SonicAxiom thank you! If I did this right, I got 50 samples of rec latency at 44,1KHz and 128 buffer size. Sounds good
Hi, thanks for this post, it's great! Though I'm having an issue. I've followed your instructions and works really well with the click track. However, when I add in my outboard gear (in this case a reverb pedal), the latency changes slightly. In fact, it works better when setting up outputs directly on the track, rather than in realnsert. Wondering if I'm doing something wrong?
Thanks for your comment, James. I can't say if you are doing something wrong (too little information). Basically, a reliable interface should always yield the same latency regardless the method you use to route signals through it. However, it's important to measure and compare the correct signal path (to avoid comparing apples with oranges)!
A reverb pedal might be a special case due to its nature. Introducing some amount of latency is part of its design. It's not a good idea to use such a device to determin the latency of a system or signal path. The added latency you are experiencing is probably caused by the pre-delay the pedal is supposed to add to the signal. If your basic latency measurement of the same signal path is correct, it' should also work correctly with the pedal inserted into it.
@@SonicAxiom Thanks! I'll do a few more tests with different pedals :)
Did it work out? My latency compensation keeps changing. I've been trying to correct this for a good while now.
Johny, what audio interface are you using?
Focusrite scarlett 18i20 usb
Windows 10
will the values be different if there were more backing tracks (drums,bass,guitars ext)? and even more if these tracks had fx on them?
no, a particular latency compensation value is only depending on the sample rate and the buffer size. Before running the latency test, you may wanna heavily stress your DAW/system with lots of plugins and tracks to find out, what a safe buffer size is for your situation without getting any audio glitches. Once this is done, run the latency test to find out the exact compensation value for that safe buffer size. If you ever change buffer size or sample rate later on, you will have to re-run the test under the new condition and will find out another latency compensation value only valid for the new conditions.
@@SonicAxiom thank you so much for this
@@yannis-sfp you're welcome, yannis!