The clicks are vertical lines on the spectrogram, (rather than the waveform).
Hi Trebor,
Paul’s L-de-clicker worked like MAGIC!! And the spectrogram?!?! Whoa! Seeing all the bits and pieces of sound that way BLEW.MY.MIND.
Forever Grateful! Thank you!
I have a 2 second playback delay issue–I have since the beginning but just learned to deal with it like having a sore tooth and chewing on the other side. I changed by Punch and Roll pre-roll to 7 sec to compensate. But it’s very annoying to deal with it every day and was wondering if you could help.
I looked up how to fix it, but only found a Forum entry from 2009 and the suggestion to do something with Audio I/O doesn’t seem to exist anymore preferences. Can you help me figure out how to deal with this?
Thanks!
For me Audcaity on Windows 8 has an annoying ~½ second delay between hitting play and playback starting.
The delay on rendering the waveform after an effect is applied is longer: ~3 seconds.
Old versions of Audacity on previous versions of Windows did not have these conspicuous delays.
I did think it could be anti-virus, but even with that disabled the delays persisted.
(IMO something wrong with the Audacity GUI, caught in an unnecessary loop, spinning it’s wheels)
Other free options are not laggy, e.g. … https://www.ocenaudio.com
Hi,
Thanks Trebor for your stories of delay. Helps to know I’m not the only one. Not sure if I’ll change DAWs quite yet. It’s taken me months to get this far along in understanding Audacity. Not ready to start at the bottom of the learning curve again just yet.
A question about crossfading clips. As a general rule, if I am using punch-copy-past with room noise between bits of audio (I use it for editing out some breaths and mouth noise at the ends of words), is there a general rule of thumb about how much loud and soft parts of the wave I should have represented in my highlighted section at the seam around the seam. Is the most seamless fade a 50/50 split, or more soft than loud or more loud than soft. I’m doing it a lot by trial and error, which may be the way all engineers do it, but I thought if I understood the concept a little better, I might be able to trial with more success than error.
Thanks!
Hi,
I just upgraded my mic system from a USB mic to a Focusrite interface and a dynamic mic with a Cloudlifter for boost. It seems that the equipment is in working order. I’ve been having troubles getting decent levels for spoken word in Audacity and someone in tech support at Sweetwater told me I wouldn’t be to use this set up with Audacity–he mentioned something about incompatible sound drivers, I think. IS THIS TRUE??? Do I have to say goodbye to this DAW? Thank you for letting me know.
Which Focusrite interface? The “Scarlett” range are popular interfaces among Audacity users.
Hi Steve,
I have a Focusrite Scarlett 8i6 3rd Gen. I can get the levels better on Reaper than Audacity from what I can tell. I don’t know Reaper well enough to be able to do a really good comparison with Audacity, but at least my gain doesn’t have to cranked up so high with Reaper to get a normal signal.
Any tips on how to configure Audacity for the new set-up?
Thanks!
The recording level has virtually nothing to do with Audacity. The only control that Audacity has on the record level is that on some machines, Audacity’s recording level slider can control the recording level slider in the Windows mixer (on computers where the recording level slider is unable to do that, it does nothing).
In all cases you can set your levels with the USB interface controls, and then with the Windows Mixer controls.
Thanks for this info. I think the problem is that the output signal from the interface isn’t being interpreted by Audacity correctly. When I checked levels on both Reaper and Audacity at the same time, both at 100% input on computer USB level (that you mentioned), Audacity registered my sound at 6dB quieter than Reaper. I sustained one note and checked between the two DAWs–Audacity showed -18db while Reaper showed -12db.
Is there anything to do about that?
6dB is a magic number. That’s half or double depending on which direction you go.
You bought a stereo interface—two sound channels. if you’re only using one of the channels, for example, to record one microphone for voice work, some systems pick that one channel and we’re good. But some systems assume you want to jam the two channels together and reduce each one by half to leave room. That’s likely where the 6dB went.
Try this. Set everything up for stereo— two channels. You should get your voice on the left and nothing on the right, but your voice should be the right volume now. No more 6dB dip.
Use the drop-down menu on the left of the track > Split stereo to mono. Delete the dead track. That will give you a proper mono voice recording.
You can decide to just go with the slightly reduced volume. Make sure the right channel volume control on the Scarlett is all the way down to suppress its noise. The Audiobook Mastering Suite of tools will make up the difference.
Did you buy a Shure SM7 to go with your Cloud Lifter? Those two products are joined at the hip for home users. Home systems feature modest volume to keep users out of overload trouble. High volume is immediately fatal for home recordings. High Quality Dynamic (moving coil) Microphones also feature modest volume and the combination isn’t the best, particularly if you’re not using a sound mixer. Enter the Cloud Lifter which is a high quality volume booster.
There are some rules with a Lifter. It must be plugged into a microphone system that supplies Phantom Power. That’s the 48v button on your Scarlett. However, it uses the phantom power up, so you can’t use a condenser microphone or any other type of microphone that also needs phantom power. Dynamic (moving coil) microphones don’t need Phantom Power, so it’s a terrific marriage.
Don’t use AIR or Pad. Pad is for playing a super loud instrument (trumpet) into the Scarlett. AIR adds distortion as a marketing ploy.
Koz
Koz,
You get a gold star!!! I have checked everywhere to figure this out and talked to numerous tech support people about my problems. I even ventured out to a Guitar Center brick and mortar store in the middle of the pandemic to try out all my equipment piece by piece to make sure nothing was broken.
Yes, the 6dB difference was the 2 outlets. Your idea to record stereo/split/delete second track did the trick.
And yes, I did get a Shure SM7B with the Cloudlifter. My impression was that the Cloudlifter would increase the gain more significantly than it has. Demos of the 2 products together showed voice actors recording at 40% of the gain knob. I’m at 60-70% of gain knob (10 o’clock). But maybe that’s normal for my interface, I don’t know. The Focusrite 8i6 got good ratings for this particular set up, so that’s why I got it. And aside from this crazy-making 6dB difference, I’m happy with it.
Thank you so very much for helping me figure this out. I was not keen on switching DAWs after putting months into learning Audacity. Thank you, as always!
I meant that my gain on this new set up is set at 2 o’clock not 10 o’clock. I must be dyslexic with analog clock faces. This gain is very different from my USB condenser that I would set at 10 or 11 o’clock. Without markings on the gain knob, I have no idea how to talk about these settings even. Hope it’s clear.
Maybe you can tell me if this is an Audacity problem or a sound system problem. The sample I’ve attached is a raw recording of the ShureSM7B with Cloudlifter. As comparison, I’ve included a sample with my USB mic.
As per your suggestion, on the Shure test, I deleted the second track on the stereo track to get accurate readings. In order to get my Shure mic to pick up my voice at usable levels (bobbing between -12 and -6), the gain on the interface is up to 2 o’clock which brings in all the electronic noise. I feel like this shouldn’t be the case. The 48v button is turned on, the interface is plugged into a power conditioner. I’m just at a loss for knowing what’s causing the dirty signal.
Thanks for your input.
Sorry, just old habit to make mp3s. Here are WAV files of what I just sent.
There are only two gain settings to worry about. All the way up and all the way down. Both of those indicate a failure, damage, or distortion somewhere.
All the rest of the swing of the dial is set with the Audacity volume indicators. The Audacity bouncing sound meter should just start turning yellow on some volume peaks or verbal emphasis. It should never get so high it turns red.
The blue waves on the timeline should sometimes, occasionally, get about half way up (0.5). Slightly over is OK.
I don’t care that the volume knob is at 10:45. Can you make Audacity volume? That’s what pays the bills.
Koz
Maybe I don’t understand.
Are you suggesting I put my interface gain up to 100% and then use the Audacity mic level to find the sweet spot for the voice?
I just did that and it seems that Audacity’s recording level slider doesn’t have any influence.
Can you hear in my SM7B recording sample what I am talking about? For that one I was recording at the level that picks up the voice with some presence without clipping, and I can’t figure out how to do it without bringing in terrible amounts of electronic noise.
I just don’t know what I’m doing wrong or what I’ve missed in the set-up process that is preventing clean sound.
The SM7B sounds very good, apart from the slight mains hum.
Mains hum can be picked-up inductively by the microphone cable, (a bit like a radio antenna).
If the mic cable is running close to mains power-cable/transformer it can pick-up hum.
So re-routing the microphone cable is worth a try, to see if that reduces the hum …
(there will be mains power cables in the wall/floor)
use the Audacity mic level to find the sweet spot
Not what I said. The Audacity indicators have to be happy. The blue waves and the jumping meter. Much of the time with USB devices the Audacity controls drop out. That leaves you with the control on your Scarlett.
SM7B_Cloudlifter_raw_WAV
The volume, presentation and in general recording is perfect. Write that down. You’re never going to hit all these specifications exactly all the time. The best you an do is get really close and let the mastering and post production tools do their thing.
However, the background noise in that sound file has a bad connection or…something. I’ll make it worse on purpose.
That’s what happens when the sound system crosses with the wall power (in the US) by accident. That can be caused by anything in the microphone pathway not being plugged in quite right to computer problems to occasionally city power.
You can listen yourself by drag-selecting part of the silence before “This is me speaking…” and listening with your speakers or headphones turned up.
I’m researching the Scarlett. Can you plug that into the wall backwards? The little power unit is two prong, not three, right?
The USB microphone in that collection doesn’t have this problem, but the SM7 is a much better microphone. We should find out why yours doesn’t work right.
the interface is plugged into a power conditioner.
Ah.
The computer and the Scarlett need to be plugged into the same thing. Try it without the conditioner. Everything should be home run into the wall. If you can’t do that, then a heavy, three-prong extension cable or power strip.
Non-fancy, please. This is one of the places you can improve a product so much it stops working. It shouldn’t have USB ports, filters, regulators, or hot and cold running beer.
Koz
The SM7B sounds very good, apart from the slight mains hum.
Correct. That’s a killer microphone.
Koz
