Recording Level Automatically Compressed - Can't Clip


I’m recording into Audacity 2.1.2 on Windows 10 Professional x64.

I’m recording a line-level source signal into the stereo L-R line-level analog input of my motherboard’s sound card (ASRock X99 Extreme4).

The Audio Host Audacity is using is MME. The recording device is “Line In (High Definition Audio)”.

No matter how high I crank the digital recording level or the analog level of the source, the input signal will not clip. The recorded waveform instead just gets more and more compressed.

I would like to disable this automatic input gain control / compressor, but can’t locate any option in Audacity or Windows that will allow me to bypass it.

Any advice?

Thank you.

disable this automatic input gain control / compressor

Probably neither one. If you actually try to make a recording that way, you will probably find it’s actually clipping.

You can get this if you cross Stereo and Mono on certain devices. Try setting up the whole chain for Stereo even if you’re trying to record a mono device or show. You never got to telling us what you’re actually recording. That can be good to know.

Audacity > Edit > Preferences > Devices > Recording.

If you cross Stereo/Mono settings by accident, Audacity may reduce the volume of the show to “make room” for work in case it gets louder…and then it never does.


Koz thank you for your reply.

I have to admit I can’t make sense of this statement:

Do you mean there is no input compression happening? I have recorded and there is no clipping. I should say, there is no hard clipping / digital clipping / signal above +1 or below -1. The waveforms appear to be getting soft-clipped in order to keep them within the +1 / -1 range. If I crank the level way beyond all reason (like maybe +10 dBu), the waveforms are smushed all to hell toward the peaks, but never they clip digitally. It looks like the maximum level is kept at about about +0.9 / -0.9 (out of +1 / -1).

I’m digitizing a record, so I have:
a stereo phono output going to a stereo phono preamp input; from the stereo phono preamp line-level output to a mixer’s stereo line-level input; and from the mixer’s line-level output to the stereo line-level input of my motherboard’s audio section.

It appears the recording settings in Audacity are for stereo, and all the recording tests I’ve done generate stereo tracks.

Since posting my original post, I’ve tested the other available Audio Hosts in Audacity, and they all appear to behave the same way. I am assuming this must be a Windows or motherboard issue at this point.

but never they clip digitally.

But the analog signal is apparently being clipped… It’s probably your soundcard. It’s actually not that unusual for the mic input on a regular soundcard to clip below 0dB, but it’s not as common with line-in.

Or, it could be something else in your analog chain. You might try it without the mixer (just because that’s an easy experiment).

If your setup is clipping at 0.9, you should be able to get good results by keeping the levels below 0.9. 90% is about -1dB, so it’s not a big difference and you need to leave some headroom for unexpected/unknown peaks anyway. (And You can always boost after recording.)

there is no hard clipping / digital clipping / signal above +1 or below -1.

That’s the digital system overloading. You can create analog overload distortion before the computer gets to it.

This is an analog microphone system overloading by adjusting the mixer wrong. The tops of the blue waves look like they’ve been clipped off.
Screen Shot 2017-02-25 at 3.53.00 PM.png
Many times it’s not that clean and obvious, but the look of the wave is unmistakable. Most times the slider for a mixer channel will have a red CLIP light to warn you the mixer is adjusted wrong.

You can see this effect by zooming into the blue waves. There are a lot of zoom tools, but I only use three.

Zoom (Windows)
– Drag-select something on the timeline and zoom into it. Control-E
– Zoom out a little bit. Control-3
– Zoom out full. Control-F

– Shift-ScrollWheel or Shift-TouchPadScrub will shift the timeline view left and right (sooner and later).

Change the timeline so it doesn’t shift by itself:
– Edit > Preferences > Tracks > [_] Update display while playing. (de-select)
Note: Update Display is good during recording.

It could be overloading anywhere.

a stereo phono preamp input

Which one?

a mixer’s stereo line-level input

Which mixer?

Does your soundcard have the usual three connections?

I have one other surround soundcard that has six different connections and I don’t think I ever sorted which one was Stereo Line-In.


The Behringer UFO-202 is a phono preamp with no adjustments. There are no reports of it ever overloading, but if it did, you would have to transfer that record some other way.


DVDdoug and Koz thank you for your responses.

The signal chain is:

Dual 1229 turntable with a Shure moving-magnet cartridge
Pro-Ject Audio “Phono Box MM” moving-magnet phono pre.
Tapco “Mix 100” mixer.
ASRock “X99 Extreme4” motherboard onboard audio input. I referred to the motherboard manual before recording. It has a surround sound audio section, and the light blue socket is the stereo line-level input socket, which is the one I’m using.

Initially, I tried feeding the phono pre’s line-level output directly into the computer audio input, but the level was way too hot and I was getting, …I guess we could call it “analog clipping” or “soft clipping”, etc. …a smashed recorded signal.

The phono pre has 40dB fixed gain.

The mixer has a red “OL” light, a yellow “+6” light, a green “0” light, and a green “-20” light. I had to slide the master fader on the mixer down from unity to -30dB in order to get an unclipped / unsmashed recorded digital signal. With the fader in that position, during recording, the -20 light on the mixer lit up pretty often, with occasional excursion up to the 0 light. The red overload light on the channel of the mixer that the phono pre’s outputs were feeding into never lit up. The audio sounded clean and undistorted [at least as clean and undistorted as 50-year-old vinyl can sound].

All of this is with “Recording Level” set to 100 in Audacity. If I plug the phono pre’s line-level outputs directly into the computer’s inputs, and turn the recording level gradually from 0 to 100 and then back down while music is recording, this is what the recorded waveform looks like:

When I zoom in on the waveform in the areas where the signal isn’t pushing up against 0 dBFS, the waveforms are clean and unclipped.
When I zoom in on the most aggressively limited/compressed area of the waveform, it looks like a “bandwidth-limited square wave” generated in Audacity: flat tops with ringing at the rising and falling edges of the plateaus.

My whole reasoning behind attenuating the analog signal, and leaving the digital recording level at maximum, is to maximize the digital resolution of the recording. The combined facts of the ringing on the clipped sections of the recordings, and of the fact that the digital recording-level control causes the recorded signal to clip or not depending on where it’s set, makes this a mystery to me.

It seems almost as if the sound card is sampling the input signal, determining that it will clip, applying some kind of processed/controlled clipping before hardening the recorded sample, and then committing the sample to the recording. This is obviously naive speculation on my part, but I can’t make heads or tails of all this behavior added together.

I think you’re losing it as you enter the soundcard.

The gain specification for the Phono Preamp is misleading. The phrase “Phono Preamp” means it knows how to untangle the RIAA distortion that the engineers built into a phonograph record. Bass notes don’t fit in a groove, so they are suppressed and treble notes are boosted to help compensate for groove noise. So the music in a groove is not flat and needs a phono preamp. There are additional exotic problems, but we keep coming back to needing to marry the cartridge to the preamp.

Moving Magnet technology is the conventional one, so the preamp doesn’t have to worry about additional boost needed for Moving Coil cartridges. I expect the preamp output to slide right into any Stereo Line-In without breathing hard. No mixer needed. That’s not what happened to you.

And this is where we sail off into the mist. You don’t have a conventional, normal soundcard, and you’re running on a Windows machine.

Windows can decide to help you out with its voice processing technology whether you want it or not. Dig in here and see if any of these settings are active.

I don’t know what to tell you about the soundcard. Did it come with a software driver or other customization settings?

When I transferred my sister’s 45s, I used a Hafler preamp. That’s a very good RIAA preamp and volume control.
Screen Shot 2017-02-25 at 21.50.53.png
and the Stereo Line-In of a Mac. Straight stereo input. I’ve used it for paid gigs forever.

I ran the Mac recording full volume and adjusted the Hafler for overall volume and overload.

And recorded in Audacity.

I played what I thought was the loudest part of the song and adjusting the Hafler for peaks about -3dB to -6dB on the Audacity sound meters and then transferred it.

I’m not sure where to go with your problem. In My Opinion you shouldn’t need the mixer in the middle, but you may depending on how the computer acts.

There are forum elves with more experience transferring vinyl into a Windows machine than I have.


The output from the pre-amp should be about right for a “line level” input.

Do you have a hi-fi amplifier? If so, try connecting the output of the pre-amp to an AUX or TAPE INPUT on the amplifier. Does it sound like you are overloading the input of the amplifier? It shouldn’t. It should sound good.

Thanks for your further replies, fellows.

I do have a hi-fi amp (AudioSource Amp One/A), with specified input sensitivity of 800 mV for full power output. When I play the record player through the phono pre directly into the amp (bypassing my usual line-level “preamp” / source selector / volume controller), the signal coming out of the speakers is extremely loud but clean and undistorted.

I just tested the phono pre’s output level by running a 6 mV, 375 Hz sine wave generated in Audacity out of my motherboard’s line-level stereo outputs into the phono pre. It output a clean sine wave at just over a volt, which puts it at about 44 dB gain at 375 Hz, which - given the RIAA curve - sounds right to me.

Taking your suggestions, Koz, I don’t see any Windows settings available to disable any strange enhancements or effects. It does seem like a motherboard issue, but going through the BIOS menus and the menus once booted, looking through the manuals, I can’t find anything. Now I’m wondering if the motherboard’s audio section is boosting the analog signal coming into it if the “Record” level on the computer is much above 20 or so. Can the digital recording level controller control an analog gain stage? Without opening the case and trying to probe the motherboard, which I am not likely to risk, I guess I’ll never know.

Either way, it may be time to start looking for a new sound card, but as long as I can set the signal to a clean level before recording, and as long as the entire signal’s not being distorted even if it’s at a low level, I guess this setup will work for now. I have to admit the recordings sound super clean on playback through Audacity.

Hopefully this wasn’t too much of a wild goose chase. I appreciate your valuable time.

So it seems that the signal from the phono pre-amp is fairly “hot”, but not unduly so. A proper “line level” input should be able to handle it, so I’m in agreement with koz that the sound card (or its settings) would appear to be the problem.

Windows will sometimes add +20 or +30 dB boost to the recording level, via a setting in the Windows Sound Control Panel. Double check that that is not occurring.

If it’s not a sound card setting, then a new sound card is probably your best solution.

We often suggest looking at the Behringer UCA-202/222 USB as it offers good sound quality at a very low price, but in this case I don’t think that will be a good choice unless you’re happy to go via your mixer. These devices do not have input level control, so they will clip if the input is too hot. The mixer will allow you to keep the signal below 0 dB so as to avoid clipping, but it’s an extra bit of kit in the signal chain. Better would be to go:
Turntable → Pre-amp → USB sound card with input level control → Computer.
Turntable → USB phono pre-amp → Computer.

I suggest USB for the interface as it does not require opening the computer case, and if you change your computer at some time you can still use the same (USB) sound card.

Behringer do make an inexpensive USB sound card with “phono” inputs (can be connected directly to the turntable), as do several other manufacturers (ART and others).