I am trying to record my vinyl record on to Audacity as a backup. I have DAC (behringer) connected to the AMP and USB connected to my Laptop
The host is Windows WASAPI, Payback Device is SPEAKER (2- USB Audio CODEC), Recording Device is SPEAKER (2- USB Audio CODEC) Loopback. But when I click record I get a red vertical line and it does not move and there is on waveform visible. I did have this working a few months ago, is it something to do with update?
This is going to record the computer’s output - which is why you are not getting anywhere.
Select the USB device without “loopback” as your recording device.
Hopefully an ADC (and probably a DAC too). 
Analog-to-digital converters are for analog input (recording). Digital-to-analog converters are for analog output (playback).
I have done this type of recording a few times. If you are using a real player with a standard output going to a usb device, you get better quality. I like your setup. Here are a few things to note:
- Behringer uses ASIO drivers, along with WASAPI; why important? With ASIO you are not limited to 48khz at 24bits most of the time. I like using at least 96khz at 32 bits BEFORE ripping it down to digital formats. I like ASIO for this reason. ASIO will support the highest rate your usb device supports. At usb2 speeds, that’s 96khz for most retail hardware (including from behringer). You may want to attempt to build Audacity with ASIO support and record from your device that way to get the higher sampling. It may sound like overkill, but I’ve done this before. I used my dad’s old record player and amp, went to an m-audio input over usb, and did the same later with a behringer over firewire (which you can now do over thunderbolt with many usb3.1 ports). WASAPI is OK, but lacks a lot of the potential of ASIO, ALSA, or COREAUDIO. ALSA is the Linux equivalent of COREAUDIO, but is a bit more limited. COREAUDIO is the APPLE audio driver package. Audacity is great for making recordings of single voice instrumentation, but without ASIO, I don’t bother using it. I built it with ASIO, then tried to use it, and did this again. The turntable went out, but what I did get was still better than a 48k. You want to capture your data at the highest sample rate you can handle, at the highest bitwidth you can handle, and then convert from there to smaller files. Otherwise, you’re only capturing a limited and thin-sounding version of the audio. Personally, I’ve been looking for a retail interface that will do DSD, which is GHZ or THZ, many thousands of times more accurate, but a single song takes a whole hard drive before converting. I wanted to see if Audacity could handle it. But all I can say about that is: A guy can dream…
- Your bitwidth may seem trivial, but it helps capture the smaller decimal differences of the sample data numbers. Some older jazz featured “less than professional” physical instruments that had a character all their own. These instruments added something to the “feel” of the music. The more accurate you are in the flaws in their frequency\key distribution, the better they will sound in your capture and on converting to other lossy formats. Distances between the datapoints will be more accurate, and the wave-form it generates will be closer; when converting, those distances will be better maintained, keeping at least their “feel” among each other, due to how the brain will be interpreting the playback. That’s why I always start with 24bits at a minimum. 16bits can work with stems that you are trying to mix, and only for getting the basic loudness distribution. When making the final passes for the mix, 24 is bare minimum, no matter what your sample rate. With a single instrument voice (mic or input with a single instrument playing into it; a single voice), low sample rates like 44.1k are actually more forgiving. Upconverting those may seem counterintuitive, but it actually EXTENDS any noticeable flaws and makes them easier to iron out with plugins like autotune. But you are dealing with pressed vinyl, so that is probably less useful. Just an FYI from my own experiences.
- Always consider your memory and your drive space. Even basic sampling rates with WAV\BWF standard files are heavier to carry than their DSD based equivalents, mostly because the external DAC handles some of the load, and they have only 5-7 bits to describe the sinewave, even though there are many more samples. This is because those bits can be sandwitched and losslessly compressed to be sent into the computer, and processed into a filespace at memory speeds much easier than with higher bitwidths with fewer samples. Playback of these files is at the same rate your system provides, but they still sound better because of their datapoint accuracy in the conversion. If you are using an SSD, I wouldn’t worry about bitwidth or sample rate. You can do up to 338khz with an i9 CPU, a decent SSD and 16gigs of memory, provided that’s the only process you are running (check startup processes and limit them----Look this up online).
As with anything, these are just my experiences. It took me some time to piece together some of the why, but it makes some sense to me. Of course, I grew up with some bad ears, and had some surgical help. Their not OEM, but they work, and once they did, I realized how awesome sound could be. Downside… …I can’t find a quiet place to sit to save my life! Everything in life has a price… …Good or bad.
You don’t need ASIO for “high resolution”.
One if it’s advantages is that it doesn’t re-sample so it doesn’t fool you into thinking you’re getting higher resolution than your hardware actually supports.
The main advantage is supposed to be low latency (low delay) so you can record yourself and monitor yourself with headphones without too much delay. But sometimes WASAPI is just as good. (WASAPI didn’t exist when ASIO was introduced.)
But when digitizing analog recordings, a few milliseconds of latency is not a concern. It’s often better to use a bigger buffer, which creates more latency, to prevent “glitches”.
Audacity doesn’t support ASIO, and some of the “cheap” Behringer interfaces don’t either.
The higher-end Behringer interfaces work with ASIO and standard Windows drivers.
For the “cheap” interfaces, Behringer offers ASIO4ALL which is like a translator or adapter that allows you to use ASIO applications… But since Audacity doesn’t support ASIO it doesn’t do anything.
And you don’t need high resolution for vinyl… “CD quality” (16-bit/44.1kHz) is WAY BETTER than analog vinyl and generally better than human hearing so usually good enough for anything.
But if your hardware supports high resolution you might as well use it. The only downside is larger files and you can down-sample later if you need/want a different “final production” format.
DSD is an odd format and not necessarily “better”. Conversion between DSD and PCM (WAV) in either direction is not lossless. But it can sound identical. And editing DSD requires special software, and if you find a DSD editor, most effect plug-ins are regular PCM.
No format (analog or digital) is “perfect” but most most common digital formats are audibly perfect.
The guys who do blind ABX tests have pretty-well demonstrated that most people can’t hear the difference between a high-resolution original and a copy down-sampled to 16/44.1.
Thanks for all the comments, very helpful.
I have change the recording option to SPEAKER (2- USB Audio CODEC) and now I get an horizontal line, no waveform displayed. I am using a Behringer UCA202 Audio in output is a USB. Is there anything else I can check?
It is possible that you have some unrelated glitch in your recording system. As a test can you record from the built-in microphone ?
Somehow this still doesn’t look like the right device to be recording from. Can you run Transport > Rescan Audio Devices, then post the results of Help > Diagnostics > Audio Device Info.
Also, it is entirely possible that as you are working on this system, you have inadvertently unplugged the wires coming from your turntable (check both ends).
The built-in mic was disabled on my Win 11 and I had the phono cables incorrectly plugged in. So working now and thanks for the tips for basic fault finding steps.
Glad to hear it is all sorted. 
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