What I learnt from this video: - Linux is actually Linux but with a longer name - unfa has a Dragonfly Reverb shirt - LSP is still awesome - Noise Repellent is still magic - Haha compressor go brrr - The Ardour MIDI tutorial is still not out - Haha multiband compressor go brrr - All of unfa's videos are awesome
The 0.1 dB peak you've got because of using 'Full' oversampling mode. The low-pass filter which is applied before down-sampling may give ringing effects which give a +0.1 dB peak at the output. To avoid this, you may just use 'Half' oversampling mode instead.
Oh, that's interesting! I must admit I didn't know the difference between the two. I should probably have read the manual. Do you think some of that information could be added as tooltips in the plug-ins?
@@unfa00 Not earlier than in 1.2.0. There's a huge refactoring and decomposition of the UI currently in progress. After the UI gets refactored, new UI features will be added.
Very timely for me. Making tons of videos for my students, and the audio is horrible. I invested in a nice microphone and it arrived the day after you posted this video. THANK YOU!!!!
Cool! I recently recorded several chapters of a book for a friend as an audio version and also stuttered a lot although I read the text before. The sentences were half a page long in some cases, so I decided to divide the long sentences into shorter ones myself. After each finished sentence, I set a marker during the recording, so that I could later better recognize where the individual parts were. I also recorded a few seconds of silence before each longer recording to better remove the noise. But I did all this with Audacity before I even knew your channel existed. In the end, I found that especially with texts that need to vary in emphasis (e.g. novels with different characters and literal speech), it is better for me to record whole chapters unedited (or almost unedited) and then listen to them several times, for example while walking. That way, mistakes in intonation can be found more easily and one learns the text by heart. In addition, it is good when speaking, if you stand and play the text as an actor. For example, in heroic passages, you might clench your fist or change the body tension. I haven't been able to finish watching your video yet, but I'll definitely do it tomorrow. Greetings from Berlin and thank you very much! 😍😁
Hi unfa, really well-done tutorial and I appreciate that you emphasized the importance of preparation and the recording environment. I'd like to add that your voice changes a lot during the course of a day, so pick a time, when you're at your best. :) It would have been nice to hear a comparison between the unprocessed recording and the final result at the end again.
Today I have learned a lot and appreciate it. I use Ubuntu, Shortcut and Gimp for my channel and really love the software. No pop-up, clicking, confusing things going arround. Sometimes I have to start my second computer with windows 10, and get constantly heart attacks while windows is updating or just showing this spinning ring without any reason. Microsoft is actually a rape of computers. :-)
BTW - Proximity effect will only work with some polar patterns... It's strongest with figure 8, cardioid works well too, but Omni won't do any bass boost.www.neumann.com/homestudio/en/what-is-the-proximity-effect
This was such a great tutorial thank you so much! I actually recorded the same script, and applied what I hoped would be equivalent filters (In premier pro) I hope I can get knowledgeable enough with audio mixing that I can make it sound as good as you!
in Russian TH-cam, Linux is not popular for media content production. I have to look for information in the foreign segment. And yes, i'm lucky with your youtube channel. Awesome lessons and tutorials. Thank you. PS.sorry for my english
In case anyone is confused by then text I'm reading - it's an old copy-pasta jokingly referring to Richard Stallman's insistance on calling Linux GNU/Linux instead.
I had two voiceover samples recorded by different people. I followed along with your video using both samples. The first sample came out GREAT.. but I couldn't get the same steps to work with the other sample. I'm very new at this, so it took a while to figure out why, and I think I got it. The first sample was recorded with a condenser mic. The second sample was recorded with a ribbon mic. So as soon as I got to the EQ stage with the ribbon mic sample, trying to use similar settings to what I used with the condenser mic sample just turned out awful. I'm not sure ribbon mic vocals need EQ at all, but I also know next to nothing about this topic. Any thoughts?
Well. I am showing you a process, but specific settings depend on what the mic profile is, what the speaker's voice is, what the room is and what the desired effect is :) I am surprised you got a good result without tweaking it to our own ear at all :) Also: the microphone I use here is a dynamic one (SM57), not a condenser one.
As an EQ Tip: you could split the spectrum int othree bands. lows (below say 300 Hz), highs (above say 1.5 KHz) and mids (everything between the other two. Then try to listen to a good sounding recording and tweak the level of each band until you get closest to the reference recording. This is EQ-matching. You probably need to train your hearing to recognize the sound of various frequencies and how to change it with EQ to get a desired effect. It takes practice.
Thank you, Unfa. I didn't know "Noise repellent". I'll certainly try. I've been using Audacity to do that part and then importing the exported wave to Ardour. By the way, when I do that some strange thing happens: my wave peaks higher when I high-pass-filter it. Does that happen with you too? Again, thank you very much. I have a cool suggestion: a tutorial to use gated reverb using Ardour. Greetings from Brazil.
Thanks! A higher peak after low pass is totally possible, as sometimes removing the high frequencies will create natural ripples of the smoother waveform, which may have a slightly higher amplitude.
@@unfa00 The same apply when you're using high-pass filter, doesn't it? When that happens do you leave as it is or do you limit to the original peak to avoid problems in the overall peak? In my case here sometimes the difference is about 2 or 3 dB.
Awesome tutorial! This was exactly what i was looking for. Im about to start voice over job,,, Do you think i could bother my neighbors (next door, upstairs, downstairs) using microphone as im living in an apartment? :/
Thanks! I think if you're going to keep the volume comparable to a regular conversation level, they shouldn't mind. If you can't hear them talking, they most likely can't hear you :)
Do you really want to monitor your audio WHILE you're recording a v.o.? I don't have any issue with latency because I don't monitor live while recording. I just do testing before hand until I am satisfied that there is no problem. Then I do the recording without headphones. I can see why I would need to monitor if I were doing music. But for just a v.o., I think it's simpler to just avoid it.
With Ardour 6.3, there is a useful "Loudness Analyzer & Normalizer" ardour.org/whatsnew.html that can automatically set the level to known platforms (TH-cam, Spotify, ...)
I believe Eq10q plugins use the GTK GUI toolkit which creates problems and collisions with Ardour - that means some versions of Eq10q won't show GUI in some versions of Ardour. I though it's a good opportunity to start using LSP plugins more.
I've challenged myself to use them as little as possible, since the DSP is sometimes of questionable quality in them. Though Calf Filter, Calf Envelope Filter, Calf Vintage Delay, Calf Stereo Tools, Calf Ring Modulator, Calf Flanger, Calf Phaser, Calf Multi Chorus are very useful and I use them whenever appropriate.
Thanks! I've heard the newest generation of Focusrite Scarlett line works great with Linux, but I haven't tested these. I use a Behringer UMC202HD. Generally I like it a lot, but sometimes I have some issues with starting JACK that won't resolve until I disconnect the interface, so I suspect it might be it's fault.
Thanks! I've corrected it now. I use Shure SM57 with the A2WS windscreen. I also use Behringer B-1 as an aux mic in the background to pick up some ambience, but it's not visible in the frame.
This video shows how to EQ the precise mic and windshield you have to sound like the legendary Shure SM7b. th-cam.com/video/s372qfTc3-I/w-d-xo.html I'm not sure, from my limited experience, most VO artists would go for such a compressed, almost robotic, sound to their work. I think I'd need to watch that again at my PC, some of the editing and post-processing whistled past me at speed!
Thanks for the link - I've checked out the video and I preferred the sound of SM57 - it's brighter. Also - did you mean that most voice over artist would go for such overexaggerated sound as I presented here, or that they wouldn't?
@@unfa00 As I understand it, they'll avoid excessive alteration of the sound. It depends on the application - some may be directed to provide a raw, edited (mistakes and noises removed) file, others, pros like Booth Junkie, just do a gate, low cut, light compression and a slight demudding with a dip in the 200-300Hz range. He does have a vocal booth so doesn't need to sweep for resonances. Audiobook narrations generally don't rely on a gate, they'll use only denoising as they need to meet a final noise floor of -60dB (ACX specification). I'm sure you know they generally avoid dynamic mics, but that's beside the point of the video. My main takeaways were the editing and some of the plugins, as Ardour is definitely not the preferred tool of most VOs! There's also Punch and Roll, which some VOs use, some don't.
Very strange. I have downloaded the project, and i have the same pluginns, but it sounds completely different than the video on youtube. For a part that is normal, but if i would describe what i hear? It seems like it is a few semitones lower in frequentie as in the video. Sort of like a small young man like you in the video, and a very big grown man from 50 years old on my system. But i do like it. It sounds very proffesional.
Yes, It is my system. I have selected all the regions and pitch-shifted them 1 semitone, and now it sounds just like in the video. I don't know why this is happening. Sorry.
@@germona You probably mixed up sampling rates. Your Ardour project should have the same sampling rate as the audio files you are importing. Ardour can convert when you import but there is the potential for error there.
@@spotlight-kyd Thanks strogon42. Because i have the same setup as Unfa (see his video on installing manjaro and all the plugins) , i have just opened the file i have downloaded from this page. When i open an existing file, i can not change the sample rate, i think. Another thing is i use Alsa and not Jack, because i cannot get it to work properly, dispite of several efforts over the years. I really hope that pipewire will take off soon.
@@germona I had a look at the Ardour project and it has indeed the wrong samplerate set. It is set to 44100 Hz, but the recorded audio is 48000 Hz. You can either start Ardour and set the audio driver to 48000 Hz and ignore Ardour's warning or open the VO.ardour file with a text editor (it is just an XML file) and change 'sample-rate="44100"'' in the second line to 'sample-rate="48000"' (without the single quotes) to fix it.
@@spotlight-kyd Thank you very much, indeed that did the trick. I did not know you could just open the ardour file in a text editor. I also did not know that the sample rate had something to do with the pitch. So learned two things today, thanks to you. God Bless you.
No. There are official builds available for as little as $1. The free demo builds go silent after 10 minutes. On Linux it's very easy to get unofficial (but still legal) builds without this demo limitation for free, though the unofficial builds often lag behind the official ones.
@@unfa00 Thanks I already bought one daw on windows, got two commerical ones (cu base and PResonus ) I was looking for a opensource DAW on linux Thanks for fast rely I am having a few issues with ARTIX + ALSA ONLY + Preonus Audiobox I might have to look into buying another interface and using the old presonus one with a raspi as a portable livestream station project instead of my main interface on my Ryzen PC Take care thanks again
Thanks for the informative video. Truthfully I didn't care for the processed audio but learned quite a bit from the steps that you took to get there. I do a podcast and use quite different techniques than you do. First, I do not like the sound of the SM 57 / 58. I the beginning I used a Blue baby bottle but as you said, room noise was a killer. Recently I switched over to an AKG C1000. I love the sound of that mic for my voice. I set the mic 12 to 18 inches away from my mouth and at a 45 degree angle to the side. As you stated this helps with all of the mouth noise that I don't want. My DAW is Harrison Mixbus. As you know it's Ardour on steroids. Mixbus gives me that old school audio sound that I love. I process with a Calf gate to get rid of noise, and use the built in Mixbus limiter, compressor, and EQ. Also use a Harrison sibilance plugin. Take a listen at Jackslife.us and give me your critique. Thanks Unfa for all of the great videos that you make.
@@ROCKHET No. I just don't remember every second of every video I made, and usually I try to spell the names of software used as I go along :) Some of the software is linked in the video description (scroll down to "LINKS"). If you need something more specific - please give me a timestamp - I can identify the plug-ins for you this way.
@@HalkerVeil Well. I almost never need to compile anything myself using Manjaro. AUR compiles things for me, it works no different than installing a regular binary package. Only sometimes it fails :D I have a video about configuring a fresh Manjaro install for audio production, maybe it'd be interesting for you? th-cam.com/video/vgrqMv3Lzfk/w-d-xo.html
@@HalkerVeil If you need help, my community chat is a good place to go - lots of people using Linux fro audio production there, and there's always someone who will help you out in trouble (when I'm not available). You can find it on chat.unfa.xyz, there's also Discord.
Completely unsolicited opinion on GNU/Linux incoming. . . That name sucks, I think GNU should be properly credited for their work on the os but the in order to become popularized you have to have a name that makes sense and linux does that, its a unix-like system where a guy named linus wrote a major component. GNU on the other hand is a three letter acronym that doesn't mean anything to most people. Imo we either have to embrace Linux as the name, or scrap both and come up with something entirely new. The gatekeeping of 'well technically its GNU/Linux' is one of last remaining reasons that people aren't switching to it. If someone says 'I was thinking of switching to linux' and you reply with that they are immediately going to both hate you and not want to switch to linux. For the sake of growth, fsf has got to give up on GNU/Linux.
I don't know if you got that, but it was supposed to be a joke. I am calling it Linux, I think saying GNU/Linux is ridiculous and counter-productive. I do respect Richard Stallman for his work and spearheading the free software movement though.
I think GNU/Hurd is a GNU system without Linux, but I've never heard about anyone actually using it. I personally gave up on the mouthful and I just say "Linux" having GNU implied.
If you're on Windows or Mac, you'd need to pay $1 for it, so yes, the official pre-compiled packages aren't free of charge. But the source code is freely available and you can compile it yourself or if you're using a Linux-based OS, you'll almost always have an unofficial package available to you free of charge. I know it's confusing, but I don't make the rules. On the other hand - is $1 too much to ask for an *optional* payment that really is needed only if you're using an expensive OS, so you probably can afford it? I just use Linux, which costs nothing, and Ardour costs nothing on it too.
Compression? COMPRESSION? Why? I mean, I know why, you just explained what it does, but WHY? Why would you ruin any audio file by applying compression? Compression should be the literally last tool in your arsenal. It should be the nuke you never want to use. ...it's going to sound bigger... my arse. It's going to ruing your range, that's what it's going to do.
Well I'm purposely overdoing the compression on this recording. You can tell I'm not using this kind of voice over sound in my videos - I just wanted to show how you can achieve this cliché effect if you'd ever need it.
@@unfa00 Yes, yes I had figured you're not using it by the fact your voice sound human. I blame the recording industry for this compression BS. The Loudness wars are REAL sheeple! It's people like you that teach well intentioned aspiring content creators how to use compression why nice music is going the way of the gramophone cylinder. Sorry, compression just triggers me. I remember this great Anime show I used to watch that had briliant music... and then the Loudness wars happened to the music they put out. I'm still angry about that... 20 years later and I'm still pissed they butchered the music in NOIR.
thx oe5RDS 77.5Khz listen on a VOICE(vocal) -tracking-ZOOM_KINECT_audio_MIC_ARRAY all you want to SAMPLErecords simple and EASY a MUSICzoomARRAY PHILIPS DATA > 8000 ;) thx
00:00 - Prerequisites/Tips for better recording
11:48 - Recording
16:57 - Editing
25:07 - Processing (various LSP Plugins in action)
43:17 - Exporting
What I learnt from this video:
- Linux is actually Linux but with a longer name
- unfa has a Dragonfly Reverb shirt
- LSP is still awesome
- Noise Repellent is still magic
- Haha compressor go brrr
- The Ardour MIDI tutorial is still not out
- Haha multiband compressor go brrr
- All of unfa's videos are awesome
The 0.1 dB peak you've got because of using 'Full' oversampling mode. The low-pass filter which is applied before down-sampling may give ringing effects which give a +0.1 dB peak at the output. To avoid this, you may just use 'Half' oversampling mode instead.
Oh, that's interesting! I must admit I didn't know the difference between the two. I should probably have read the manual. Do you think some of that information could be added as tooltips in the plug-ins?
@@unfa00 Not earlier than in 1.2.0. There's a huge refactoring and decomposition of the UI currently in progress. After the UI gets refactored, new UI features will be added.
Very timely for me. Making tons of videos for my students, and the audio is horrible. I invested in a nice microphone and it arrived the day after you posted this video. THANK YOU!!!!
Awesome, good luck with the new recordings!
Thanks Unfa, I enjoyed that. It’s always fun to watch someone else’s workflow.
Good presentation, now I learned many new things about voiceovers. thanks for the video.
Thank you so much for the new video! Very high quality. Stylish, useful. It's nice to watch and listen. (With respect)
Thank you! I'm glad you're liking it :)
@@unfa00 And thank You for the high-quality You Tube channel!
Cool! I recently recorded several chapters of a book for a friend as an audio version and also stuttered a lot although I read the text before. The sentences were half a page long in some cases, so I decided to divide the long sentences into shorter ones myself. After each finished sentence, I set a marker during the recording, so that I could later better recognize where the individual parts were. I also recorded a few seconds of silence before each longer recording to better remove the noise. But I did all this with Audacity before I even knew your channel existed.
In the end, I found that especially with texts that need to vary in emphasis (e.g. novels with different characters and literal speech), it is better for me to record whole chapters unedited (or almost unedited) and then listen to them several times, for example while walking. That way, mistakes in intonation can be found more easily and one learns the text by heart.
In addition, it is good when speaking, if you stand and play the text as an actor. For example, in heroic passages, you might clench your fist or change the body tension.
I haven't been able to finish watching your video yet, but I'll definitely do it tomorrow. Greetings from Berlin and thank you very much! 😍😁
Great informative comment.
Ahh! I've been waiting for this sort of video for so long!! So excited to watch
Hi unfa, really well-done tutorial and I appreciate that you emphasized the importance of preparation and the recording environment. I'd like to add that your voice changes a lot during the course of a day, so pick a time, when you're at your best. :) It would have been nice to hear a comparison between the unprocessed recording and the final result at the end again.
Today I have learned a lot and appreciate it. I use Ubuntu, Shortcut and Gimp for my channel and really love the software. No pop-up, clicking, confusing things going arround. Sometimes I have to start my second computer with windows 10, and get constantly heart attacks while windows is updating or just showing this spinning ring without any reason. Microsoft is actually a rape of computers. :-)
BTW - Proximity effect will only work with some polar patterns... It's strongest with figure 8, cardioid works well too, but Omni won't do any bass boost.www.neumann.com/homestudio/en/what-is-the-proximity-effect
I didn't know that! Thanks for mentioning this :)
This was such a great tutorial thank you so much! I actually recorded the same script, and applied what I hoped would be equivalent filters (In premier pro) I hope I can get knowledgeable enough with audio mixing that I can make it sound as good as you!
Thanks! Practise makes perfect! Good luck :)
Very very pro and usable!
in Russian TH-cam, Linux is not popular for media content production. I have to look for information in the foreign segment. And yes, i'm lucky with your youtube channel. Awesome lessons and tutorials. Thank you.
PS.sorry for my english
Best tutorial ever ❤️✨
Thanks for the help Unfa!
In case anyone is confused by then text I'm reading - it's an old copy-pasta jokingly referring to Richard Stallman's insistance on calling Linux GNU/Linux instead.
I had two voiceover samples recorded by different people. I followed along with your video using both samples. The first sample came out GREAT.. but I couldn't get the same steps to work with the other sample. I'm very new at this, so it took a while to figure out why, and I think I got it. The first sample was recorded with a condenser mic. The second sample was recorded with a ribbon mic. So as soon as I got to the EQ stage with the ribbon mic sample, trying to use similar settings to what I used with the condenser mic sample just turned out awful. I'm not sure ribbon mic vocals need EQ at all, but I also know next to nothing about this topic. Any thoughts?
Well. I am showing you a process, but specific settings depend on what the mic profile is, what the speaker's voice is, what the room is and what the desired effect is :)
I am surprised you got a good result without tweaking it to our own ear at all :)
Also: the microphone I use here is a dynamic one (SM57), not a condenser one.
As an EQ Tip: you could split the spectrum int othree bands. lows (below say 300 Hz), highs (above say 1.5 KHz) and mids (everything between the other two. Then try to listen to a good sounding recording and tweak the level of each band until you get closest to the reference recording. This is EQ-matching. You probably need to train your hearing to recognize the sound of various frequencies and how to change it with EQ to get a desired effect. It takes practice.
This is a very significant video. Thanks.
Thanks for the the tips! Great video!
Thanks! You're welcome :)
Thank you, Unfa. I didn't know "Noise repellent". I'll certainly try. I've been using Audacity to do that part and then importing the exported wave to Ardour.
By the way, when I do that some strange thing happens: my wave peaks higher when I high-pass-filter it. Does that happen with you too?
Again, thank you very much. I have a cool suggestion: a tutorial to use gated reverb using Ardour.
Greetings from Brazil.
Thanks! A higher peak after low pass is totally possible, as sometimes removing the high frequencies will create natural ripples of the smoother waveform, which may have a slightly higher amplitude.
@@unfa00 The same apply when you're using high-pass filter, doesn't it? When that happens do you leave as it is or do you limit to the original peak to avoid problems in the overall peak? In my case here sometimes the difference is about 2 or 3 dB.
Awesome tutorial! This was exactly what i was looking for. Im about to start voice over job,,, Do you think i could bother my neighbors (next door, upstairs, downstairs) using microphone as im living in an apartment? :/
Thanks! I think if you're going to keep the volume comparable to a regular conversation level, they shouldn't mind. If you can't hear them talking, they most likely can't hear you :)
@@unfa00 interesting! Great to know. thank you :)))
Do you really want to monitor your audio WHILE you're recording a v.o.? I don't have any issue with latency because I don't monitor live while recording. I just do testing before hand until I am satisfied that there is no problem. Then I do the recording without headphones. I can see why I would need to monitor if I were doing music. But for just a v.o., I think it's simpler to just avoid it.
My tips: cut down on coffee, drink enough water
You mean: for doing voice overs?
Hi, what is the name of this software?
Please cubase 5 recording software link
ok Unfa. it sounds quite good. the bass is too loud in some places. i like natural sound. it depends from the context
Thank you🐧
Thank you.
hey man how do I install the
Noise Repellent plug-in, in windows. thank you
As far as I am aware, that plugin is not available for Windows. Maybe someone could port it over? I don't know.
@@unfa00 alright man, thank you
Thankyou ☺️👍
Unfa, you're fucking awesome
With Ardour 6.3, there is a useful "Loudness Analyzer & Normalizer" ardour.org/whatsnew.html that can automatically set the level to known platforms (TH-cam, Spotify, ...)
Thank you for your great videos. Is there a reason why you are not using eq10q anymore?
I believe Eq10q plugins use the GTK GUI toolkit which creates problems and collisions with Ardour - that means some versions of Eq10q won't show GUI in some versions of Ardour. I though it's a good opportunity to start using LSP plugins more.
@@unfa00 ah, ok. thank you :)
did you break up with calf plugins?
I've challenged myself to use them as little as possible, since the DSP is sometimes of questionable quality in them. Though Calf Filter, Calf Envelope Filter, Calf Vintage Delay, Calf Stereo Tools, Calf Ring Modulator, Calf Flanger, Calf Phaser, Calf Multi Chorus are very useful and I use them whenever appropriate.
Nice video any suggestion on a good audio interface that works well under linux
Thanks! I've heard the newest generation of Focusrite Scarlett line works great with Linux, but I haven't tested these. I use a Behringer UMC202HD. Generally I like it a lot, but sometimes I have some issues with starting JACK that won't resolve until I disconnect the interface, so I suspect it might be it's fault.
Thank you
Have you tried Ocenaudio?
I haven't, but it doesn't seem to be open-source so it's outside of my scope of interest :)
Im just curious where did u use blender it the video?
For the 3D intro animation, and the outro screen :)
@@unfa00 oh nice 👍😘
can you pls advice the name or brand of the best microphone for recording voice overs
Could be Rode NT1-A. I'm using a dynamic mic so it captures less background noise in non-ideal recording environment.
@@unfa00 thank you so much 🌹
The link to adour is wrong. What Microphone do u use?
Thanks! I've corrected it now.
I use Shure SM57 with the A2WS windscreen.
I also use Behringer B-1 as an aux mic in the background to pick up some ambience, but it's not visible in the frame.
0:20 - richard stallman intensifies.
multiplier effect on stallman
This video shows how to EQ the precise mic and windshield you have to sound like the legendary Shure SM7b. th-cam.com/video/s372qfTc3-I/w-d-xo.html
I'm not sure, from my limited experience, most VO artists would go for such a compressed, almost robotic, sound to their work.
I think I'd need to watch that again at my PC, some of the editing and post-processing whistled past me at speed!
Thanks for the link - I've checked out the video and I preferred the sound of SM57 - it's brighter.
Also - did you mean that most voice over artist would go for such overexaggerated sound as I presented here, or that they wouldn't?
@@unfa00 As I understand it, they'll avoid excessive alteration of the sound. It depends on the application - some may be directed to provide a raw, edited (mistakes and noises removed) file, others, pros like Booth Junkie, just do a gate, low cut, light compression and a slight demudding with a dip in the 200-300Hz range. He does have a vocal booth so doesn't need to sweep for resonances. Audiobook narrations generally don't rely on a gate, they'll use only denoising as they need to meet a final noise floor of -60dB (ACX specification).
I'm sure you know they generally avoid dynamic mics, but that's beside the point of the video.
My main takeaways were the editing and some of the plugins, as Ardour is definitely not the preferred tool of most VOs!
There's also Punch and Roll, which some VOs use, some don't.
Very strange. I have downloaded the project, and i have the same pluginns, but it sounds completely different than the video on youtube. For a part that is normal, but if i would describe what i hear? It seems like it is a few semitones lower in frequentie as in the video. Sort of like a small young man like you in the video, and a very big grown man from 50 years old on my system. But i do like it. It sounds very proffesional.
Yes, It is my system. I have selected all the regions and pitch-shifted them 1 semitone, and now it sounds just like in the video. I don't know why this is happening. Sorry.
@@germona You probably mixed up sampling rates. Your Ardour project should have the same sampling rate as the audio files you are importing. Ardour can convert when you import but there is the potential for error there.
@@spotlight-kyd Thanks strogon42. Because i have the same setup as Unfa (see his video on installing manjaro and all the plugins) , i have just opened the file i have downloaded from this page.
When i open an existing file, i can not change the sample rate, i think.
Another thing is i use Alsa and not Jack, because i cannot get it to work properly, dispite of several efforts over the years. I really hope that pipewire will take off soon.
@@germona I had a look at the Ardour project and it has indeed the wrong samplerate set. It is set to 44100 Hz, but the recorded audio is 48000 Hz. You can either start Ardour and set the audio driver to 48000 Hz and ignore Ardour's warning or open the VO.ardour file with a text editor (it is just an XML file) and change 'sample-rate="44100"'' in the second line to 'sample-rate="48000"' (without the single quotes) to fix it.
@@spotlight-kyd Thank you very much, indeed that did the trick. I did not know you could just open the ardour file in a text editor. I also did not know that the sample rate had something to do with the pitch. So learned two things today, thanks to you. God Bless you.
Is ardour cripplewere ?
Free downloadable copy mentions it kills itself after 10 minutes
No. There are official builds available for as little as $1. The free demo builds go silent after 10 minutes. On Linux it's very easy to get unofficial (but still legal) builds without this demo limitation for free, though the unofficial builds often lag behind the official ones.
@@unfa00 Thanks I already bought one daw on windows, got two commerical ones (cu base and PResonus )
I was looking for a opensource DAW on linux
Thanks for fast rely
I am having a few issues with ARTIX + ALSA ONLY + Preonus Audiobox
I might have to look into buying another interface and using the old presonus one with a raspi as a portable livestream station project instead of my main interface on my Ryzen PC
Take care
thanks again
unfa it's exactly as TeaBoi say: "All of unfa's videos are awesome"!!! :-)!
Thanks for the informative video. Truthfully I didn't care for the processed audio but learned quite a bit from the steps that you took to get there. I do a podcast and use quite different techniques than you do. First, I do not like the sound of the SM 57 / 58. I the beginning I used a Blue baby bottle but as you said, room noise was a killer. Recently I switched over to an AKG C1000. I love the sound of that mic for my voice. I set the mic 12 to 18 inches away from my mouth and at a 45 degree angle to the side. As you stated this helps with all of the mouth noise that I don't want. My DAW is Harrison Mixbus. As you know it's Ardour on steroids. Mixbus gives me that old school audio sound that I love. I process with a Calf gate to get rid of noise, and use the built in Mixbus limiter, compressor, and EQ. Also use a Harrison sibilance plugin. Take a listen at Jackslife.us and give me your critique. Thanks Unfa for all of the great videos that you make.
What is name of the software?
The DAW is called Ardour.
Nice one!
Thanks!
hi i'm trying to follow along. what software are you using ?
Didn't I name the software I used here?
So the point of your reply is to be petty instead of answering the question?. Ight. Noted.
@@ROCKHET No. I just don't remember every second of every video I made, and usually I try to spell the names of software used as I go along :)
Some of the software is linked in the video description (scroll down to "LINKS").
If you need something more specific - please give me a timestamp - I can identify the plug-ins for you this way.
May I ask if you’re German?
I'm Polish :)
"As close as possible without touching it."
Just like a stripper.
Now I have to install a whole NEW type of Linux just to get it to work right...
This is why I don't use Linux anymore.
Could you elaborate? What's the problem?
@@unfa00 Oh nothing. Just remembering how I had to compile every program to work on every other linux distro so it works correctly.
@@HalkerVeil Well. I almost never need to compile anything myself using Manjaro. AUR compiles things for me, it works no different than installing a regular binary package. Only sometimes it fails :D
I have a video about configuring a fresh Manjaro install for audio production, maybe it'd be interesting for you?
th-cam.com/video/vgrqMv3Lzfk/w-d-xo.html
@@unfa00 Hmm maybe I'll check it out and see what's changed.
@@HalkerVeil If you need help, my community chat is a good place to go - lots of people using Linux fro audio production there, and there's always someone who will help you out in trouble (when I'm not available). You can find it on chat.unfa.xyz, there's also Discord.
Completely unsolicited opinion on GNU/Linux incoming. . . That name sucks, I think GNU should be properly credited for their work on the os but the in order to become popularized you have to have a name that makes sense and linux does that, its a unix-like system where a guy named linus wrote a major component. GNU on the other hand is a three letter acronym that doesn't mean anything to most people. Imo we either have to embrace Linux as the name, or scrap both and come up with something entirely new. The gatekeeping of 'well technically its GNU/Linux' is one of last remaining reasons that people aren't switching to it. If someone says 'I was thinking of switching to linux' and you reply with that they are immediately going to both hate you and not want to switch to linux. For the sake of growth, fsf has got to give up on GNU/Linux.
I don't know if you got that, but it was supposed to be a joke. I am calling it Linux, I think saying GNU/Linux is ridiculous and counter-productive. I do respect Richard Stallman for his work and spearheading the free software movement though.
👌✌️👍
Ok!
I would like to see a GNU system running without Linux
(this whole discussion about gnu linux is pedantic AF)
and yes, I user Fedora Linux :)
I think GNU/Hurd is a GNU system without Linux, but I've never heard about anyone actually using it. I personally gave up on the mouthful and I just say "Linux" having GNU implied.
It is not a free software!!!
If you're on Windows or Mac, you'd need to pay $1 for it, so yes, the official pre-compiled packages aren't free of charge.
But the source code is freely available and you can compile it yourself or if you're using a Linux-based OS, you'll almost always have an unofficial package available to you free of charge.
I know it's confusing, but I don't make the rules. On the other hand - is $1 too much to ask for an *optional* payment that really is needed only if you're using an expensive OS, so you probably can afford it?
I just use Linux, which costs nothing, and Ardour costs nothing on it too.
@@unfa00 yes, i have Windows. Is 1 dollar every month?
IIRC it's one-time if you just want the current version, and monthly if you want to have access all the time.
Suiiii
Compression? COMPRESSION? Why? I mean, I know why, you just explained what it does, but WHY? Why would you ruin any audio file by applying compression? Compression should be the literally last tool in your arsenal. It should be the nuke you never want to use. ...it's going to sound bigger... my arse. It's going to ruing your range, that's what it's going to do.
Well I'm purposely overdoing the compression on this recording. You can tell I'm not using this kind of voice over sound in my videos - I just wanted to show how you can achieve this cliché effect if you'd ever need it.
@@unfa00 Yes, yes I had figured you're not using it by the fact your voice sound human. I blame the recording industry for this compression BS. The Loudness wars are REAL sheeple!
It's people like you that teach well intentioned aspiring content creators how to use compression why nice music is going the way of the gramophone cylinder.
Sorry, compression just triggers me. I remember this great Anime show I used to watch that had briliant music... and then the Loudness wars happened to the music they put out. I'm still angry about that... 20 years later and I'm still pissed they butchered the music in NOIR.
linux
thx oe5RDS 77.5Khz listen on a VOICE(vocal) -tracking-ZOOM_KINECT_audio_MIC_ARRAY all you want to SAMPLErecords simple and EASY a MUSICzoomARRAY PHILIPS DATA > 8000 ;) thx
to produce a good track do not record crap
Please cubase 5 recording software link