Sorry for the late upload on this one folks! Been dealing with some glitchy Premiere Pro renders ever since updating to Premiere Pro 23. See around 18:37 in the video for a visual of the weird render issues I've been having. If any Premiere Pro wizards have experienced this and know a solution, please let me know. Seems to happen once in every render I do lately. I'll be rolling back to version '22 for part #14!
I saw that glitch. For a minute I thought I was watching my cable company. I have Premier, FCP and Resolve. They all work pretty much the same - just gotta find where each task is located in the workspace. Thanks for all the videos! 👍🏼
At 11:30 of this video, you directed the users to go to "Mix", "Pre-Fader Metering". I can't believe how long I have been researching why my recordings are clipping while the audio levels all looked solid. Since I could not find the answer in any of my searches, I decided to just watch your sequence of tutorials. Yep - at 11:30 in lesson #13 I got an answer I could not find in any of my searches. Finally - my recorded wav files are no worse than me! Many thanks!
Thanks Josh for this series. I've been using Logic for some time now but not really learned all the features of it. I'm Swedish and must say You're a good teacher, I can follow You easily not beeing natively speaking English and You are accurate and pedagogical in your way of showing us poor watchers on how to use Logic to the fullest. Thanks from cold Sweden.😊
First of all, Thank you Josh for your knowledge shared here with so many content! You are amazing, and the way you explain is absolutely brilliant! I just would like to ad one little thing about the audio format; the AIFF format is only for Apple users and the WAVE format is for any user Windows or Apple, etc. Hope this could be useful for someone, because back in the day one of my first projects couldn’t be done with another engineer because of the format incompatibility. Thank you again! 😊🙏🏻✨
Absolutely amazing! Thank you for your contribution to my unfinished (no, unstarted)) education that I am now trying to get by myself. The best info on Logic!
Loving the series. I’m currently on #11 but skipped ahead to #13 to see if you deal with headphone sends etc. When I couldn’t find it in this video or the descriptions of later videos, I looked in your older series and found something (#66?). Will you be covering this in the new series - how to control sends to monitors and headphones separately. Best to set up a separate bus, separate outputs etc? Thx!!
I'm learning a lot from your videos. I am wanting to record our a cappella quartet. I used to be familiar with ProTools but no longer. Just purchased Logic Pro
Wonderful series. Thank you so much for putting these together. I'm learning so much. As for your premeire issues. Switch to Resolve. I dropped Adobe 2 years ago and have not looked back. I know thats not helpful but honestly it's far superior in my opinion and i've been editing for 30 years. Fusion has a bit of a learning curve but I can tell you will pick it up quick. Good luck.
I own Final Cut Pro, but I still need to sit and learn how to use it. I've considered Resolve as well. But again, gotta find time to properly use it. I think the issue here is just version 23 of Premiere, this never happened in version 22.
@@MusicTechHelpGuy FCP is a good option for you too. It’s fairly easy to use too. Better options for you and a lot more stable on rendering. I’m sure your Adobe issue is just a glitch but who needs those right? Best of luck. Keep ‘em coming. I look forward to them. And again, a huge thank you.
Loving this series; thank you! So, when recording acoustic drums with two mics (overhead and kick), should the inputs be set to mono? Also, you kinda lost me with the pre-fader gain thing. Is there another video that covers this? Thanks!
Thank you for the video. What thickness are the strings you use on your acoustic guitar? And what brand of strings do you use? Thank you for your answer.
I know it's been a while since you recorded this and I appreciate the solid tutorial on recording with a microphone. Now I've moved to logic 11 and it seems like the fader does not change the level of the recording input whether it's in pre-fader mode or not. fyi
Thank you for excellent video. I have two questions to whoever can answer. 1. Why does the volume meter stop at whatever value it peaked at? I do I reset it? I thought it followed along and showed the played volume at the present moment. 2. Don't understand Pre Fader. What does it do?
If you’re recording by yourself, and you can’t reach your keyboard, use the Logic Remote app on your iPhone or iPad. I record at home, and most of the time I’m able to sit at my desk (for recording guitar or bass). If I’m doing my own vocals or keyboards I use the Logic Remote on my iPhone.
Logic Remote rocks. I use it all the time! I actually had a blurb about it at the beginning of this video, but I cut it for time, but I plan on doing a flyover of Logic Remote for this series, fairly soon. Thanks.
Initially thanks for this content, I have been following you for a few years. I have a problem where I keep hearing pops (not the proximity effect low end pops or ones caused by bad editing) after I use Flex Time etc. I have my buffer down low for recording to reduce latency. However I read that could be contributing as even though I take the buffer back up afterwards, as it was recorded so low I may have been stressing the CPU which can cause pops. Could this be true? Also is there any real downside to recording in low latency mode apart from some plug ins and processes are bypassed to reduce CPU usage?
ok - dumb question. Sometimes, say when recording drums, my audio interface volume needs to be turned WAY down to get a signal that's not clipping. Is that the best way to do it - just turn the audio interface volume knob down to 2-3? Or should I get a decibel reduction XLR adapter to plug the mic into? Just turning down certainly works fine, but wasn't sure if that affects the overall sound or signal. No pad button in Logic, I don't think (I'm using a Focusrite Clarett 8 pre - no pad button there either). Thoughts? Thanks!
Turning the gain way down is totally normal. Especially for close mics like kick and snare. You can also try angling the mics a bit so they are not perpendicular to the drum heads, if you’re not already doing that. This will reduce the proximity effect a bit, resulting in slightly less bass energy = lower signal. You might also check the focusrite control app, there might be a software pad in the app.
Wondering is it the same thing duplicating the guitar track and panning left and right as recording another take on another track and panning left and right? Any preference?
Duplicating a guitar track and panning will do nothing for the stereo image, it will only make the recording louder. It's the minor differences between the left and right channels that create the stereo width. So when I do rhythm guitar parts, I always double the guitar with a second recording, then pan L and R.
is an audio interface REQUIRED to connect my microphone to Logic? I just bought and xlr cable that connects straight to my macbook. Thank you for the input!!
Normally you need an audio interface or a USB microphone. Sounds like your XLR to USB cable might have a built in converter? Because one of the things the interface does is convert analog audio to digital. If the XLR-USB shows up as an input device, then you should be good.
The combined signal of my guitars, plus the click track are clipping the output. Fortunately this has no effect on the recording signal and the clipping is gone as soon as you turn off the metronome.
Please respond! If you are using direct monitoring while recording, wouldn't latency be irrelevant? So, couldn't you set the buffer size to 512 when recording?
Yep! If your audio interface has direct monitoring, you don't need to worry about latency or changing the buffer. Just make sure to mute your channel you are recording on, and also, you will not be able to hear any effects that are running live on that channel. But, yes, if you use direct monitoring, you essentially have no perceivable latency. Because the direct monitor signal is coming straight from the interface to your HF, and never hits Logic.
Can’t make my new condenser mic record anything. I have Komplete Audio 2, with phantom power, XLR3 to XLR3. I’m selecting the correct input. Obviously i’m missing something. Anyone can help with this?
Regarding sample rates; You can find a number of videos where they performed null tests to show that recording higher than 48 kHz does not improve audio quality. In fact higher sample rates like 192 kHz introduce aliasing artifacts. Personally I use 44.1 kHz for everything. Digital audio only requires 2 samples per cycle to faithfully reproduce any waveform. So for 44.1 kHz you will reproduce frequencies up to 22 kHz, which you cannot hear. A big misconception is that digital audio records stair steps. It does not. That would require infinite bandwidth and slew rate. It will ALWAYS reproduce a smooth sine wave. It can’t not reproduce a smooth sine wave. All AD/DA converters have brick wall filters to limit the upper frequency response to prevent aliasing. So recording at 192 will not give you a higher frequency response or better definition (since you only need two samples per cycle… so a 20 kHz sine wave only needs 40 kHz sampling rate. The extra 4.1 kHz is to compensate for the low pass brick wall filter). You will not get better high frequency response at higher sample rates. Sorry this part of this video is incorrect.
Thanks for the detailed response, this is a lot to unpack, and there are a number of factors here. I've been a recording engineer for almost 20 years. Like I said in the video, there are many other factors that affect audio quality other than sample rate and bit depth. Most of those tests you are referring to are just capturing audio A/D, then spitting back out again D/A. Not actually working with the audio in a practical setting. There's a lot more to sample rates than AD/DA. Anti-alising filters work better at higher sample rates as well. Anti-aliasing filters don't actually filter down to (-) infinity. There is still room for aliasing and foldover harmonics to come through the filter -- I've heard it, and seen it myself on several occasions. The quality of the converter also matters here. It's why my Symphony I/O sounds fantastic at 44.1 kHz on cymbals, but a Focusrite Scarlett sounds gummy and mushy in the top end (also the analog front-end matters a lot here). I agree 100% that using higher sample rates just for recording quality is a waste. I use 44.1 kHz most of the time as well. However, keep in mind that the sample rate also affects the operating frequency of the DAW, so when you process the signal, you are processing at the sample rate you set when you starting making your recordings -- to me, this is an even more important reason to consider using a higher sample rate. Again, not required, but a reason to "consider". Processing and effects can introduce additional harmonics above 22 kHz in the DAW. Not to mention any sound sources that are originated in the DAW do no hit the anti-aliasing filter on input -- like software synthesizers. A third reason to use a higher sample rate would be time and pitch manipulation. Typically using a higher sample rate will result in less artifacts produced from heavy time and pitch manipulation. So while what you're saying is absolutely true, it's not all about recording quality, often it's about processing quality within the DAW. Which is why in the video I give as sparse of an explanation as I can, because this is still a beginner video, and I could go on for days about sample rate and bit depth. Thanks.
@@MusicTechHelpGuy You won’t hear anything above 22 kHz because of the anti aliasing low pass filter on the converter. And that high frequency stuff can fold back as aliasing artifacts. Present Day Productions has a good video on this. Plus humans can’t hear that high. I’ve been a recording engineer about 40 years. I enjoy your channel. I started with DAWs with Deck II in 1995 and switched to Cubase in ‘98. Used Cubase and Peo Tools until about 6 years ago when I switched to Logic full time. With the way Apple hides features I still pick up good hints from your videos!
@@DavidRavenMoon Lower sample rates absolutely can still result in audible foldover harmonics in many situations. The anti aliasing filter on input is not applied to sound sources and processing produced within the DAW. Pull up a synthesizer like the ES2 with a basic sawtooth at 44.1 kHz. Play a really high note, and look at the frequency spectrum in the channel EQ. You'll see the foldover in the EQ. You'll be able to hear it too. It gums up the top end. Then switch the project over to 96 kHz for example. Play the same note -- foldover gone. So again, my point is that there is way more to this than just AD/DA.
I know this request is absolutely terrible, and the worst idea for a video, but I do not have an actual microphone, but I love to pull samples for house music just with the iPhone microphone when I’m on iOS garage band, now that I’m on logic I could use the MacBook microphone, but if there is a good way to set up a sampling chain to avoid unnecessary artifacts from the background noise, and finding a good distance to have the MacBook in reference to the audio sample, that would be super awesome to see! 😅🙏
I've actually used the "Voice Memos" app on my iPhone to capture samples. It's actually pretty cool, and you can get some interesting sounds -- and the extra background noise actually can add the vibe of the instrument you are creating with the samples. Perhaps I'll do a video on this in the future. Thanks.
@@MusicTechHelpGuy thanks for that! The extra white noise has definitely helped give me some fun results, just recording parts of Spider-Man movies through the speaker can give the sample a cool gritty feeling. It can just be hard to work with some of the artifacts sometimes 😅
I’m using apogee duet with new Logic Pro X. My apogee control shows all audio is right but when I record no audio records on the tracks. It’s not on mute. Has anyone else had this issue???
Can you explain the difference between take folders and Track Alternates? I’ve heard that Track Alternates are the equivalent of the Playlist feature in ProTools
We'll get to track alternatives in the near future. They def are similar to playlists in PT, but not really that efficient for comping. More like you can save 2 or more different versions of a recording on the same track, and switch between them. As another commenter noted above, they can be very helpful for saving your comped take on say "Alternative A" and then the flattened and merged comp on "Alternative B". But there are a lot of great uses for them. Track > Show Track Alternatives.
@@MusicTechHelpGuy Thank you! I’ve taught Logic at the college level and used it since 2009 but it seems like Track Alternatives is one of those features that I didn’t touch that often because I’d use Quick Swipe Comping all the time. Thanks!
@@pianomaniac14 Same here. I've barely used them, but I can definitely see how they can be helpful. I too have been teaching music and audio production in college for about 12 years.
@@MusicTechHelpGuy You’re lucky. I keep trying to pitch production and computer music programs to colleges that don’t have one. I’m a recent graduate of Peabody Conservatory’s Computer Music Masters program and I built and taught a synth theory and electronic music course for one semester but then they ghosted me afterwards. This field is so tough! I’m still job hunting and everyone seems full.
For some reason whenever I record and listen to the play back, it is slightly delayed. It's fine when I'm monitoring, there is negligible latency. I don't know why this started happening. My solution for now is just to record a take, then drag the recording to the left by a few milliseconds to compensate for the delay. Not the biggest deal but it is annoying and takes me even more out of the creative mindset.
Check the recording delay in the Logic Pro > Preferences/Settings > Audio. It's a slider underneath the Buffer Size. You might have accidentally slid that away from zero.
Sorry for the late upload on this one folks! Been dealing with some glitchy Premiere Pro renders ever since updating to Premiere Pro 23. See around 18:37 in the video for a visual of the weird render issues I've been having. If any Premiere Pro wizards have experienced this and know a solution, please let me know. Seems to happen once in every render I do lately. I'll be rolling back to version '22 for part #14!
I saw that glitch. For a minute I thought I was watching my cable company. I have Premier, FCP and Resolve. They all work pretty much the same - just gotta find where each task is located in the workspace. Thanks for all the videos! 👍🏼
At 11:30 of this video, you directed the users to go to "Mix", "Pre-Fader Metering". I can't believe how long I have been researching why my recordings are clipping while the audio levels all looked solid. Since I could not find the answer in any of my searches, I decided to just watch your sequence of tutorials. Yep - at 11:30 in lesson #13 I got an answer I could not find in any of my searches. Finally - my recorded wav files are no worse than me! Many thanks!
Thanks Josh for this series. I've been using Logic for some time now but not really learned all the features of it. I'm Swedish and must say You're a good teacher, I can follow You easily not beeing natively speaking English and You are accurate and pedagogical in your way of showing us poor watchers on how to use Logic to the fullest. Thanks from cold Sweden.😊
First of all, Thank you Josh for your knowledge shared here with so many content! You are amazing, and the way you explain is absolutely brilliant! I just would like to ad one little thing about the audio format; the AIFF format is only for Apple users and the WAVE format is for any user Windows or Apple, etc. Hope this could be useful for someone, because back in the day one of my first projects couldn’t be done with another engineer because of the format incompatibility. Thank you again! 😊🙏🏻✨
Absolutely amazing! Thank you for your contribution to my unfinished (no, unstarted)) education that I am now trying to get by myself. The best info on Logic!
This series is really amazing! I've been watching, practicing and following along every day. I have found it to be very helpful.
This was great. Especially the improvisation in the end❤❤❤❤
Great video! You always make it so easy to follow. Nice playing Josh 😊
Great information, so important to know most of this things, thanks!
I must say you got me back into Logic Pro, been away from it since Covid you doing a wonderful job keep up the good work much respect.
Very informing, partly review, partly new to me, but very helpful. Looking forward to #14! 👍🏻👏🏻🙏🏻😊
I wish I would have watched this series when I first started using logic! I've learned so much. Thank you!
Loving the series. I’m currently on #11 but skipped ahead to #13 to see if you deal with headphone sends etc. When I couldn’t find it in this video or the descriptions of later videos, I looked in your older series and found something (#66?). Will you be covering this in the new series - how to control sends to monitors and headphones separately. Best to set up a separate bus, separate outputs etc? Thx!!
Youre the absolute GOAT for theeseee, thank you!!
I'm learning a lot from your videos. I am wanting to record our a cappella quartet. I used to be familiar with ProTools but no longer. Just purchased Logic Pro
It was worth waiting for. Excellent as always.
Freaking love it bro!! Thank you so much !!
Great Teaching for all your videos
U forgot to talk about the low latency mode .. One of the best features in logic recording stuffs.
Again, a very informative and practical video. Love your work In delivering such great content thank you
Hi, could you explain why you need 2 guitar tracks playing the exact same thing? I love this tutorial, thanks so much!!!
Music Tech Help Guy University 🙌
Wonderful series. Thank you so much for putting these together. I'm learning so much. As for your premeire issues. Switch to Resolve. I dropped Adobe 2 years ago and have not looked back. I know thats not helpful but honestly it's far superior in my opinion and i've been editing for 30 years. Fusion has a bit of a learning curve but I can tell you will pick it up quick. Good luck.
I own Final Cut Pro, but I still need to sit and learn how to use it. I've considered Resolve as well. But again, gotta find time to properly use it. I think the issue here is just version 23 of Premiere, this never happened in version 22.
@@MusicTechHelpGuy FCP is a good option for you too. It’s fairly easy to use too. Better options for you and a lot more stable on rendering. I’m sure your Adobe issue is just a glitch but who needs those right? Best of luck. Keep ‘em coming. I look forward to them. And again, a huge thank you.
Thanks again for another great tutorial!
Wow.Thank You.
Loving this series; thank you! So, when recording acoustic drums with two mics (overhead and kick), should the inputs be set to mono?
Also, you kinda lost me with the pre-fader gain thing. Is there another video that covers this? Thanks!
Brilliant, nice tune too,
what if your using a USB mic? I have one but I can't hear the audio and vocals together in the headphones
AMAZING GUITAR
Thank you for the video. What thickness are the strings you use on your acoustic guitar? And what brand of strings do you use? Thank you for your answer.
I know it's been a while since you recorded this and I appreciate the solid tutorial on recording with a microphone. Now I've moved to logic 11 and it seems like the fader does not change the level of the recording input whether it's in pre-fader mode or not. fyi
well done, thanks
Thank you for excellent video. I have two questions to whoever can answer.
1. Why does the volume meter stop at whatever value it peaked at? I do I reset it? I thought it followed along and showed the played volume at the present moment.
2. Don't understand Pre Fader. What does it do?
Thanks Josh!!
If you’re recording by yourself, and you can’t reach your keyboard, use the Logic Remote app on your iPhone or iPad.
I record at home, and most of the time I’m able to sit at my desk (for recording guitar or bass). If I’m doing my own vocals or keyboards I use the Logic Remote on my iPhone.
Logic Remote rocks. I use it all the time! I actually had a blurb about it at the beginning of this video, but I cut it for time, but I plan on doing a flyover of Logic Remote for this series, fairly soon. Thanks.
I wasn't aware of the app. Downloading now. Thanks
Apart from being a Logic Pro scientist you also play guitar very nicely, mate.
Very interesting thank you for the video 🎼👌
Initially thanks for this content, I have been following you for a few years. I have a problem where I keep hearing pops (not the proximity effect low end pops or ones caused by bad editing) after I use Flex Time etc. I have my buffer down low for recording to reduce latency. However I read that could be contributing as even though I take the buffer back up afterwards, as it was recorded so low I may have been stressing the CPU which can cause pops. Could this be true? Also is there any real downside to recording in low latency mode apart from some plug ins and processes are bypassed to reduce CPU usage?
ok - dumb question. Sometimes, say when recording drums, my audio interface volume needs to be turned WAY down to get a signal that's not clipping. Is that the best way to do it - just turn the audio interface volume knob down to 2-3? Or should I get a decibel reduction XLR adapter to plug the mic into? Just turning down certainly works fine, but wasn't sure if that affects the overall sound or signal. No pad button in Logic, I don't think (I'm using a Focusrite Clarett 8 pre - no pad button there either). Thoughts? Thanks!
Turning the gain way down is totally normal. Especially for close mics like kick and snare. You can also try angling the mics a bit so they are not perpendicular to the drum heads, if you’re not already doing that. This will reduce the proximity effect a bit, resulting in slightly less bass energy = lower signal. You might also check the focusrite control app, there might be a software pad in the app.
Wondering is it the same thing duplicating the guitar track and panning left and right as recording another take on another track and panning left and right? Any preference?
Duplicating a guitar track and panning will do nothing for the stereo image, it will only make the recording louder. It's the minor differences between the left and right channels that create the stereo width. So when I do rhythm guitar parts, I always double the guitar with a second recording, then pan L and R.
is an audio interface REQUIRED to connect my microphone to Logic? I just bought and xlr cable that connects straight to my macbook. Thank you for the input!!
Normally you need an audio interface or a USB microphone. Sounds like your XLR to USB cable might have a built in converter? Because one of the things the interface does is convert analog audio to digital. If the XLR-USB shows up as an input device, then you should be good.
Hi , I have version 10.7.4 and it is not able to show all takes at the same time for editing. Only can switch between them . What I am doing wrong ?
Hi , can you do a video on how to record and mix Saxophone in Logic. Thanks
Thanks for the great tutorials! I have one question: Why is during the recording the stereo out red and during playback green? All the best Dirk
The combined signal of my guitars, plus the click track are clipping the output. Fortunately this has no effect on the recording signal and the clipping is gone as soon as you turn off the metronome.
@@MusicTechHelpGuy Thanks a lot, I have never understood why the output is clipping. Cheers from Germany
Please respond! If you are using direct monitoring while recording, wouldn't latency be irrelevant? So, couldn't you set the buffer size to 512 when recording?
Yep! If your audio interface has direct monitoring, you don't need to worry about latency or changing the buffer. Just make sure to mute your channel you are recording on, and also, you will not be able to hear any effects that are running live on that channel. But, yes, if you use direct monitoring, you essentially have no perceivable latency. Because the direct monitor signal is coming straight from the interface to your HF, and never hits Logic.
Superb series 👍
Can you make a video on recording an a cappella quartet?
Can you explain what pre metering is?
Can’t make my new condenser mic record anything. I have Komplete Audio 2, with phantom power, XLR3 to XLR3. I’m selecting the correct input. Obviously i’m missing something. Anyone can help with this?
Regarding sample rates; You can find a number of videos where they performed null tests to show that recording higher than 48 kHz does not improve audio quality. In fact higher sample rates like 192 kHz introduce aliasing artifacts.
Personally I use 44.1 kHz for everything.
Digital audio only requires 2 samples per cycle to faithfully reproduce any waveform. So for 44.1 kHz you will reproduce frequencies up to 22 kHz, which you cannot hear.
A big misconception is that digital audio records stair steps. It does not. That would require infinite bandwidth and slew rate. It will ALWAYS reproduce a smooth sine wave. It can’t not reproduce a smooth sine wave.
All AD/DA converters have brick wall filters to limit the upper frequency response to prevent aliasing. So recording at 192 will not give you a higher frequency response or better definition (since you only need two samples per cycle… so a 20 kHz sine wave only needs 40 kHz sampling rate. The extra 4.1 kHz is to compensate for the low pass brick wall filter).
You will not get better high frequency response at higher sample rates. Sorry this part of this video is incorrect.
Thanks for the detailed response, this is a lot to unpack, and there are a number of factors here. I've been a recording engineer for almost 20 years. Like I said in the video, there are many other factors that affect audio quality other than sample rate and bit depth. Most of those tests you are referring to are just capturing audio A/D, then spitting back out again D/A. Not actually working with the audio in a practical setting. There's a lot more to sample rates than AD/DA. Anti-alising filters work better at higher sample rates as well. Anti-aliasing filters don't actually filter down to (-) infinity. There is still room for aliasing and foldover harmonics to come through the filter -- I've heard it, and seen it myself on several occasions. The quality of the converter also matters here. It's why my Symphony I/O sounds fantastic at 44.1 kHz on cymbals, but a Focusrite Scarlett sounds gummy and mushy in the top end (also the analog front-end matters a lot here). I agree 100% that using higher sample rates just for recording quality is a waste. I use 44.1 kHz most of the time as well. However, keep in mind that the sample rate also affects the operating frequency of the DAW, so when you process the signal, you are processing at the sample rate you set when you starting making your recordings -- to me, this is an even more important reason to consider using a higher sample rate. Again, not required, but a reason to "consider". Processing and effects can introduce additional harmonics above 22 kHz in the DAW. Not to mention any sound sources that are originated in the DAW do no hit the anti-aliasing filter on input -- like software synthesizers. A third reason to use a higher sample rate would be time and pitch manipulation. Typically using a higher sample rate will result in less artifacts produced from heavy time and pitch manipulation. So while what you're saying is absolutely true, it's not all about recording quality, often it's about processing quality within the DAW. Which is why in the video I give as sparse of an explanation as I can, because this is still a beginner video, and I could go on for days about sample rate and bit depth. Thanks.
@@MusicTechHelpGuy You won’t hear anything above 22 kHz because of the anti aliasing low pass filter on the converter. And that high frequency stuff can fold back as aliasing artifacts. Present Day Productions has a good video on this. Plus humans can’t hear that high.
I’ve been a recording engineer about 40 years. I enjoy your channel. I started with DAWs with Deck II in 1995 and switched to Cubase in ‘98. Used Cubase and Peo Tools until about 6 years ago when I switched to Logic full time. With the way Apple hides features I still pick up good hints from your videos!
@@DavidRavenMoon Lower sample rates absolutely can still result in audible foldover harmonics in many situations. The anti aliasing filter on input is not applied to sound sources and processing produced within the DAW. Pull up a synthesizer like the ES2 with a basic sawtooth at 44.1 kHz. Play a really high note, and look at the frequency spectrum in the channel EQ. You'll see the foldover in the EQ. You'll be able to hear it too. It gums up the top end. Then switch the project over to 96 kHz for example. Play the same note -- foldover gone. So again, my point is that there is way more to this than just AD/DA.
@@MusicTechHelpGuy we will have to agree to disagree on this.
th-cam.com/video/hs1On87Ixe4/w-d-xo.html
th-cam.com/video/-jCwIsT0X8M/w-d-xo.html
I know this request is absolutely terrible, and the worst idea for a video, but I do not have an actual microphone, but I love to pull samples for house music just with the iPhone microphone when I’m on iOS garage band, now that I’m on logic I could use the MacBook microphone, but if there is a good way to set up a sampling chain to avoid unnecessary artifacts from the background noise, and finding a good distance to have the MacBook in reference to the audio sample, that would be super awesome to see! 😅🙏
I've actually used the "Voice Memos" app on my iPhone to capture samples. It's actually pretty cool, and you can get some interesting sounds -- and the extra background noise actually can add the vibe of the instrument you are creating with the samples. Perhaps I'll do a video on this in the future. Thanks.
@@MusicTechHelpGuy thanks for that! The extra white noise has definitely helped give me some fun results, just recording parts of Spider-Man movies through the speaker can give the sample a cool gritty feeling. It can just be hard to work with some of the artifacts sometimes 😅
hey are you in located Michigan ?
I’m using apogee duet with new Logic Pro X. My apogee control shows all audio is right but when I record no audio records on the tracks. It’s not on mute. Has anyone else had this issue???
Isn't there a setting where if the Record button is engaged you can hear the audio and skip having ti engage the Input switch?
If I turned off the input how do I get it to show me the inputs again?
Can you explain the difference between take folders and Track Alternates? I’ve heard that Track Alternates are the equivalent of the Playlist feature in ProTools
We'll get to track alternatives in the near future. They def are similar to playlists in PT, but not really that efficient for comping. More like you can save 2 or more different versions of a recording on the same track, and switch between them. As another commenter noted above, they can be very helpful for saving your comped take on say "Alternative A" and then the flattened and merged comp on "Alternative B". But there are a lot of great uses for them. Track > Show Track Alternatives.
@@MusicTechHelpGuy Thank you! I’ve taught Logic at the college level and used it since 2009 but it seems like Track Alternatives is one of those features that I didn’t touch that often because I’d use Quick Swipe Comping all the time. Thanks!
@@pianomaniac14 Same here. I've barely used them, but I can definitely see how they can be helpful. I too have been teaching music and audio production in college for about 12 years.
@@MusicTechHelpGuy You’re lucky. I keep trying to pitch production and computer music programs to colleges that don’t have one. I’m a recent graduate of Peabody Conservatory’s Computer Music Masters program and I built and taught a synth theory and electronic music course for one semester but then they ghosted me afterwards. This field is so tough! I’m still job hunting and everyone seems full.
once upon a time the first thing you learned in a studio was how to make a good cup of tea. Then if you were lucky you'd learn to plug in a mic.
For some reason whenever I record and listen to the play back, it is slightly delayed. It's fine when I'm monitoring, there is negligible latency. I don't know why this started happening. My solution for now is just to record a take, then drag the recording to the left by a few milliseconds to compensate for the delay. Not the biggest deal but it is annoying and takes me even more out of the creative mindset.
Check the recording delay in the Logic Pro > Preferences/Settings > Audio. It's a slider underneath the Buffer Size. You might have accidentally slid that away from zero.
automatic crossfade not working for me 😭
Last time I tried recording audio without a microphone I found it quite difficult. (it's a joke, people!)
👍🇩🇪
I still can't make a good beat