Sound Card Reviews

i bought a soundcard from “trust” today. The 5.1 External Surround Sound Card SC-5500p
(510EX 5.1 Sound Expert External USB). Works very good with audacity. Is very cheap, no drivers needed!

Note added by Gale Andrews: This card is also an inexpensive way of adding “stereo mix” ability to computers where the built-in sound device does not support this. I found drivers were needed on Windows Vista and 7 to get stereo mix ability. I also found the Windows 7 driver does not install correctly on Windows 7 X64, but the Vista driver does and works well.

Asus Xonar Essence STX
124 dB SNR / Headphone Amp PCIe card for Audiophiles!
Ins and outs:
Inputs: 1x 6.30mm jack (Shared by Line-In/Mic-In)
Outputs: 2x RCA jack (Front R / Front L); 1x 6.30mm jack (Headphone out); 1x High-bandwidth Coaxial/TOS-Link combo port supports 192KHz/24bit

  • really good and amazingly quiet headphone output
  • very good shielding and high-quality components
  • works in linux (still WIP)
  • 3 selectable impedance ranges for the headphone’s output (up to 600 ohm)
  • easily exchangeable opamps (but I can’t find a good reason to do it)
  • shared mic-in / line-in (I’d rather have two separate inputs)
  • at the time of this writing, some features still not supported in the linux driver yet (for example: can’t select if the input is line-in or mic-in)

If you’d like to use your high-end headphones to listen to music from you computer this is THE soundcard!
Works fine in Audacity both for recording (could only test in windows due to linux driver limitations) and for playback (tested in linux and windows)

Highly recommended!

ART USB Dual Pre
Dual-mic preamp which can be used as external usb soundcard too.
Ins and outs:
Inputs: 2x XLR / 1/4-inch TRS Combi jack balanced/unbalanced
Outputs: 2x 1/4-inch TRS; 1x 1/8-inch TRS headphone

  • no drivers needed, works out-of-the-box in Linux (Debian, running kernel 2.6.32), Windows (XP Pro) and MacOS-X (Snow Leopard)
  • can be powered from usb, external power supply or 9V battery (or a combination of all these)
  • rather small size compared to what it has to offer
  • headphones monitor output can be used to play sound from the computer through USB.
  • very low noise for the price tag
  • can provide phantom power to the mics
  • if I find one I’ll update this post (some ppl complain about the size of the 9V battery pocket, I haven’t tried it so can’t complain about it…)

Works fine in Audacity (1.3.11-beta) for both recording and playback in Linux, MacOS-X and Windows (tested in all of the 3 OS’s).

Highly recommended![/quote]

Thanks for your reviews, bgravato.

I see fairly regular criticisms on Windows that the output of Art USB devices is “too hot”. Inexperienced users may want to consider the level of the input into the ART and the level of the system input. See this ART FAQ for Windows 7 (and Vista).


Shure Brothers X2U

This is a single, straight-line path sound channel from professional XLR-type microphone to the USB connection in your computer. You can read about it in the web pages. It’s pretty amazing, but it does have a couple of shortcomings.

– As usual with USB equipment, you can’t get very far from the computer, usually about 3M without inviting data errors and odd digital noises in the show. USB is not a long-haul system. The up side, of course, is that a top quality, rubber coated XLR cable can go a hundred feet without significant damage.

– The X2U doesn’t have a lot of gain. I’ve never used it where I had to turn it down, but many times I’ve wanted to turn it up. I did some real world sound tests with a commercially available phantom-powered microphone and it held its own against larger mixers as long as you didn’t need very much umph.

I announced a short track with the mic in the X2U and then did a similar track with the mic plugged into a Peavey PV6. The X2U was cranked all the way up and produced a comfortable track. The Peavey still had over 30dB of boost left to add if needed – the faders were only half-way up and the microphone trim was at 3/4.

The X2U doesn’t need any fancy software; it looks like a standard audio device. Plug it in and go. I didn’t try it on Windows. It has provision to play computer music into its own headphone socket, so if you did it right, you could do a complete sound-on-sound session. It’s 16-bit. I used it both at 48000 and 44100.

You can’t tell from the sales illustrations how small this thing is…

Behringer UCA 202

Manufacturers Specification:

2 channel (stereo in/out)
Line level in/out
16 bit 44.1/48kHz

Connectors RCA, unbalanced
Input impedance approx. 27 kW
Max. input level 2 dBV

Connectors RCA, unbalanced
Output impedance approx. 400 W
Max. output level 2 dBV

Socket Toslink, optical cable
Output format S/PDIF

Socket 1/8" TRS stereo jack
Output impedance approx. 50 W
Max. output level -2 dBu, 2 x 3,7 mW @ 100 W

USB 1.1 type A

Frequency response:
10 Hz to 20 kHz, ± 1 dB @ 44.1 kHz sample rate
10 Hz to 22 kHz, ± 1 dB @ 48.0 kHz sample rate

0.05 % typ. @ -10 dBV, 1kHz

-77 dB @ 0 dBV, 1 kHz
A/D 89 dB typ. @ 1 kHz, A-weighted

Signal-to-noise ratio
D/A 96 dB typ. @ 1 kHz, A-weighted

USB connection 5V 100 mA max

I have a very old version of this device - the label has fallen off and there is no headphone or S/PDIF socket.

Performance testing using:
Acer 5735 laptop
Using USB 2.0 port
Linux Ubuntu 9.04 operating system
(Results identical whether running from mains power or batteries.)
Audacity 1.3.13 alpha

Noise level is -78.3 dB with the input level set at 100%
This gives a 0.4 dB drop if I link the output directly into the input.
Testing the frequency response - flat (within 0.1 dBfs) from 20Hz to 19kHz (-2.5 dB at 20kHz)

Latency correction:
Pulse Audio/ALSA
Recording quality 44100kHz 32 bit stereo
234 milliseconds

Jack Audio (rt = off)
(safe settings)
Recording quality 44100kHz 32 bit stereo
49 milliseconds

(fastest settings)
Recording quality 48000kHz 32 bit stereo
36.7 milliseconds

Additional information for installing and running Audacity Daily build on Ubuntu 9.04:

This is the set-up that I use, - hopefully it will work for others:

If not already installed, install PulseAudio device chooser, and PulseAudio Volume Control.
PulseAudio Device chooser will load into the panel and allow quick convenient access to PulseAudio Volume control.

Set Audacity to use Host=ALSA and Playback/recording devices to “default”.
This should configure Audacity correctly to use your default sound system settings.
The input/output (recording/playback) devices can then be set using PulseAudio Volume Control.

The latest Audacity 1.3.12 is available from here:

To use Audacity with Jack - ensure that you have a working/stable Jack system running before opening Audacity.
Open Audacity and set the Host preference to Jack (Edit > Preferences > Devices)
Set the recording sample rate the same as Jack is using. (Edit > Preferences > Quality)
Exit preferences. Everything should be ready to go.
Audacity will use the Jack system inputs and outputs by default.

[Update] I tried connecting the output of a hi-fi CD player directly to the line input of the UCA-202. No matter what settings I used on the computer the signal was always clipped (distorted) a bit during loud music. I would not recommend using the UCA-202 in this way. Provided that the device that is connected to the UCA-202 has an output level control there is no problem.

I’m searching for a plain, analog sound card that isn’t too dreadful. It needs to have Stereo Line-In, Stereo Line-Out and/or Headphone-Out. It needs to have a Mic-In with 20dB Mic Boost. This is the spec that kills you when you’re Googling. Nobody ever writes down the boost tool.

I have a Creative SB Live! that fills the bill, but I’m not shocked that it hasn’t been made since dirt was new.

There is a really terrible sound card at work that fills the bill, too, but that manufacturer has the reputation of producing trash – both hardware and sound.

It’s going in our Linux machines to record live capture of training sessions – either to accept a mixing desk stereo Line-Out, or direct plug of a microphone. The catch is it has to work with our highly custom software, drivers, capture software, and hardware configurations. Every time I suggest a USB solution, the programmers and support people flee. They’re willing to not flee if I can plug in a relatively plain analog sound card.

I ran across a Blaster Audigy SE card, but I can’t tell if it has a microphone boost.


The Quick Start manual is available here:
(not sure if the link will work as they use dynamic web pages). The manual is in .CHM format so you will need Windows (or Wine on Linux) to read it.
It says that the blue socket is “Line IN/Mic In/Digital I/O”
As with the old SBLive cards I would not expect the microphone input to be very good quality, but probably better than the average laptop PC.

I couldn’t find the instruction book at all, so this is a big help even though I’m going to need to scare up a Windows machine to read it.

I did run across a comment that all Blaster cards have Microphone Boost on the chipset, but elect to, or not to, support it in software. Mostly not, and there followed pages of work-arounds, since most people found, as I did, that the microphone connection is useless without it.

This is why our normal mode is to use an external mixing desk and go in to the Line-In of the sound card. Unfortunately, since the corporate move, we’re facing putting some twenty of these in service. Mixing desk prices rise rapidly even with a cheap desk.


I believe most motherboard’s built-in soundcards have mic boost, whether it’s hardware or software boost beats me…

I’m on the laptop now, so I can’t confirm, but I believe my asus xonar essence stx has mic boost option and it works on linux, but probably is also a bit over your desired budget…

Some years ago there was a chaintech soundcard which had very good reviews and it was very cheap (about $25 USD in the USA). I only didn’t buy one because I couldn’t find it for sell in europe, and ordering it from the usa would make it 3x or so more expensive. I dunno if they still make it…

How can you judge bang for the buck in a consumer audio card (before you buy it)?

Apart from price, and which jacks are there for ins and outs, what are the key questions that a ‘home user’ with an interest in high fidelity recording (as distinct from a professional audio engineer) might want answered when deciding on a sound card for analog – digital conversion?

Most cards have an analog pre-amplifier stage before the ADC. Like any other analog amplifier, this stage may have limitations in frequency response, distortion and crosstalk. If you buy a hi-fi amplifier you probably check these things. Unfortunately these are rarely specified for sound cards. They can only be measured using fancy test equipment, so they are rarely tested on ‘user reviews’.

The other area where a music fan is likely to hear a difference is in the signal to noise ratio. That is the range (in dBFS) from the electronic noise floor of the card to the level at which your musical peaks are going to be captured in a program like Audacity. This S/N ratio depends on:

(1) The electronic noise floor in the sound card. You can measure this using Audacity. To see the advantage of a good 24-bit card, I think it is necessary to record in 24-bit (soon to become possible in PortAudio / Audacity for Windows we are promised). At 16 bit there is no resolution between -90 and -96 dbFS.

(2) The ability of the sound card to take your input signal strength without analog clipping. This is hard to tell, but you may get some idea from the specified nominal input levels of the sound card and the specified nominal output level of your source device (microphone or amplifier line out). The trouble is that these specifications come in many different units (dbU, dbV, mV RMS, etc). These may be mathematically convertible but then you need to consider the impedance specifications – now you need to become a professional audio engineer! Another method, after you have paid up for a card, is to look for suspicious plateaus on peaks in the recorded waveform.

(3) The ability of the sound card to convert your input signal peaks to 0 dBFS (or just below) for recording. If you record peaks at a lower level they can be amplified later, but the noise floor will be amplified in proportion, so (ignoring effects like noise removal which may introduce their own distortion problems) the S/N ratio is set when you record. You can exert some control over this (trading headroom for S/N ratio) by using the input volume control on the sound card (if one exists) or in the recording software. The Audacity input volume slider which is coupled to Windows 7 input volume level for the selected sound card.

(4) Maybe whether the sound card can work with your software for 24-bit recording, so that you are actually getting something better than 1-2 bit recording at low volumes assuming your entire audio chain has a noise floor below -80 dBFS. For more about this, see

So … it would be great if sound card reviews at least specified the noise floor. Stevethefiddle did this (and more) for the UCA202 review above. There has to be some agreed standard way specify the level, because there are always a few outlying samples and there can be some weird effects of DC offset on waveforms around the noise floor level.

For example:

(i) Eliminate known sources of electrical hum or noise from the vicinity of your computer and sound card (e.g. noisy power supplies, mobile phones).

(ii) Select the Recording Device in Audacity and set the input level slider at full scale (1.0 in Audacity = 100 in Windows Sound Level).

(ii) Record for 5 seconds with no input cable attached to the sound card.

(iii) Select the entire recording. Apply Effect > Normalize > Remove DC offset (once only).

(iv) Expand the waveform display vertically and examine it in dB view while zoomed into the noise level and the 5 second time window.

(v) Note the dB level above which there are fewer than ten samples per second (out of 44.1 kHz sampled).
This is a measure of the noise floor from the sound card. Do this in a separate recording for any altered settings (such input slider level), and note these settings along with the corresponding noise floor.

Model: Your inbuilt notebook soundcard.

It may be all you need to convert your songs from cassette or vinyl into MP3s!


On a desktop computer you may get Line in and Microphone jacks, but on a notebook there will probably only be a 1/8” microphone jack.

Don’t despair (yet). In many recent notebooks the inbuilt audio device uses a high definition (HD) audio codec, the ‘microphone’ jack is wired as a TRS stereo jack, and it is capable of being configured to accept either microphone or line signal levels. BUT, whether that hardware capability is accessible depends on the device driver; which is commonly configured and supplied only by the notebook manufacturer. For example, devices like the IDT 92HD75B have the full capability but IDT does not supply drivers to consumers, that is strictly up to the computer manufacturer. The Windows 7 generic HD audio drivers do not have the capability to alter this configuration. You need to check if it is present in the superimposed driver for your specific machine. This should be accessible through Windows control panel, for example as “IDT audio control panel” or another name matching the manufacturer (Intel, Realtek, Soundmax, …). If your notebook manufacturer did not bother to include the capability to select line versus microphone input levels, all you can do is ask them to include it in a driver update. Lots of luck with that.

The HD audio codec supported by these inbuilt devices may go all the way up to stereo 24 bit 192000 Hz (Studio Quality). Look under Windows Sound > Recording Devices > Microphone Properties > Advanced > Default Format. But it is very unlikely that you will really get Studio Quality from these devices (which wholesale for about $1 each in lots of 10,000). More likely you will push the limits of your CPU in processing without any benefit over 16 bit 44,100 Hz (CD Quality) because the electronic noise level from these integrated devices is way above -96 dBFS.

Of course the noise level from your well-loved (and used) cassettes, vinyl records, and associated consumer pre-amplifier stage is likely also way above -96 dBFS. The noise floor is determined by the noisiest link in the audio chain, so there may be nothing to gain by spending on a more expensive sound card. Use Audacity to check the noise levels from the audio device alone, then from a quiet section of your source recording (between song tracks) and decide for yourself.

Typically a 1/8” stereo headphone jack.

You can’t beat the price.

You can not use the hardware capability to deliver stereo line in signal to the ADC unless your notebook manufacturer includes the configuration setting in the OEM driver implementation.

The noise floor for these devices is typically around -48 dBFS with the Windows input sound level controller set to 100 (full scale) and no ‘microphone boost’. This is likely to be close to the required level for line-in signals around a nominal -10 dBV (316 mV RMS). The noise level may drop to around -71 dbFS with the Windows input sound level controller set to 0. But any input signal that requires a zero input volume setting will likely be clipping in the analogue stage of the audio device, before the ADC step. See this post for an example of the effects on waveforms:

Griffin iMic (now iMic 2)

A popular, low-priced USB device that is about equal in sound quality to the inbuilt audio device in many recent notebooks – with the ability to switch between stereo microphone and line signal strengths that is not provided in some notebook audio drivers.

1/8” TRS jack, switchable for stereo microphone or line level input strength

1/8’ TRS stereo headphone jack

Computer Connection:

Available for Windows XP and Mac. Not claimed to support Windows Vista or 7, but it can be configured after automatic detection simply by setting to 2 channel 16 bit 44,100 Hz (CD Quality) under Windows Sound > Recording Devices > Microphone Properties > Advanced > Default Format. This seems a little strange, as this Windows setting is ostensibly about ‘shared mode’ (and Audacity 1.3.12 uses either MME or DirectSound emulated by Win 7; not yet WASAPI which includes shared or exclusive modes of interaction with the audio hardware). But pragmatically it does the trick.

Cheap, compact.

The iMic is marketed without specs and without model numbers that clearly reflect design changes – so what you purchase may not behave like the unit that someone else reviewed.

For the iMic 2 that I tested:

With the input switch at the Line position, the noise floor is around -48 dBFS with the Windows input sound level controller set to 100 (full scale). This is likely to be close to the required level for line-in signals around a nominal -10 dBV. The noise is around -70 dbFS with the Windows input sound level controller set to 0, but you are unlikely to have a source signal appropriate to that setting.

With the input switch at the Mic position, the noise floor is around -49 dBFS at input volume 100 and -71 dBFS at input volume 0. It can be tempting to use the iMic at 0 input volume in the Mic position with a (consumer -10 dBV) line level signal for low noise. Unfortunately this will likely give clipping in the analogue stage of the audio device, before the ADC step. Such clipping may not show on the Audacity clipping alerts; you have to look carefully at the peak waveforms. See this post for an example:

Apparently the iMic uses a good quality ADC, but it has cut corners on noise reduction to keep the price down. Beware nearby sources of hum and electrical noise. See

If you have a low-noise signal source, you will benefit from spending a few extra dollars on a lower-noise device like the Behringer UCA202 reviewed above.

Roland (Cakewalk) UA-1G

I focussed on use to transcribe from vinyl to digital.
There is a thorough review about recording from guitar input at

Manufacturers Specification:

2 channel (stereo in/out at analog line level and optical digital, plus headphone out)
1 channel microphone and guitar in

Advanced (ASIO) driver On: 24 bit 32/44.1/48/96 kHz
Advanced Driver Off: 16 bit 32/44.1/48 kHz

Twin RCA jacks, nominal input level -10 to +4 dBu

GUITAR IN (switchable to MIC IN)
1/4” phone jack, nominal input level -30 to -16 dBu

MIC IN (monaural dynamic type)
1/4” phone jack, nominal input level -40 to -26 dBu

MIC IN (plug-in powered monaural miniature condenser type)
1/8” powered phone jack; can be used simultaneously with the ¼” guitar/mic in jack

DIGITAL IN (shared jack with powered mic in)
Optical mini type conforms to IEC60958

Twin RCA jacks
Nominal output level -10 dBu

1/8" TRS stereo jack
adjustable volume knob

DIGITAL OUT (shared jack with phones out)
Optical mini type conforms to IEC60958

USB 1.1 type A

USB connection 5V 200 mA

Input level dial (for all analog inputs)
Phones volume dial
Switch for guitar/mic level from shared jack
Switches for sample rate, play or record at 96 kHZ, input monitor (below 96 kHz), analog or digital input, advanced driver on/off.

USB (red LED for power from the USB port)
IN (green LED flashes above a moderate analog signal level to ADC)
OUT (green LED flashes above a moderate analog signal level from DAC)
PEAK (red LED flashes “if audio input signal exceeds the allowable level”??)

Performance tested using:

HP mini 5102 (Intel Atom N450 1.66GHz, X-25M G2 SSD, 2GB RAM)
USB 2.0 port
Windows 7 pro 32 bit
Audacity 1.3.12 beta

Noise levels (tested by recording with no input connections at 24 bit in Sonar LE then examining zoomed waveforms in Audacity) were around -78 dBFS with the UA-1G input level knob set at 100% (5 o’clock), -86 dBFS at 2 o’clock, and -89 dBFS at 0% (7 o’clock).

Set-up per manufacturer instructions (with Advanced Driver installed and switched on) gives Default Format = 2 channel 16 bit 44,100 Hz (CD Quality) under Windows Sound > Recording Devices > Microphone Properties > Advanced. Cakewalk Technical Support advised that this setting applies only to use in shared mode (simultaneous use by multiple applications). I changed it to 24 bit in any case.

The owner’s manual states for each analog input to “Adjust the input level knob until the level is as high as you can get it without causing the PEAK indicator to light”. I tried this with various input signal strengths. With music at nominal 350 mV (-9 dbV) from my Rotel RA-1312 consumer pre-amp it needed the input level dial at about 2 o’clock. With higher input signals (up to nominal 1.5 V from an FM receiver) it required correspondingly lower settings on the dial. But in every case, recording this way (with software input level set at 100%) in either Audacity or Sonar LE always gave the peaks of the recorded waveform at -10 dBFS. I emailed Cakewalk Technical Support several times to try to establish exactly what are the triggers for the peak indicator. I asked explicitly whether it is intended to indicate clipping at the analog input stage or clipping at the digital output stage or some defined (-10 dBFS) headroom. I explained that I was trying to establish whether it was sensible to ignore this peak indicator and set levels based on the recorded waveform when users did not want to trade S/N ratio for 10 dBFS of headroom (for example when transcribing to digital from finished vinyl records). I did not get an answer about the exact triggers, but the advice was

“This is a good question. The peak indicator on the UA-1G is more of a warning signal than a traditional peak light.

Basically, the UA-1G is geared toward beginners; the average consumer. It was made to not peak or clip, which is why there is a maximum output of -10 dBu on the device. The only way your signal would sound distorted when coming through the UA-1G is if the signal coming into the device was already super hot and over driven.

So yes, you can sensibly ignore the peak indicator on the UA-1G in this case. For more accurate dBu levels, we recommend devices like the UA-25EX which has better preamps and a higher output level than the

That is confusing to me, as the peak indicator has nothing to do with the output line level (-10 dBu). So I gave up on the emails (no offense, compared to some other companies Cakewalk were great to reply at all and the replies were relevant if slightly vague; but it is a noreply system so you have to jump through hoops again on the Cakewalk web site every time you would like clarification). Instead I ran a test for clipping when recording the same track (from the Thelma Houston Sheffield Lab LP via my Rotel RA-1312 pre-amp line out) with:

  1. The UA-1G peak indicator just off (which required the UA-1G input level knob at 2 o’clock and gave recorded peaks near -10 dBFS)
  2. The recorded waveform peaks around -1 dBFS (which required the UA-1G input level knob at full scale and illuminated the UA-1G peak indicator much of the time).
    In both cases, the Audacity = Windows input level slider was left at 100%. From the zoomed waveforms (note the different vertical zoom levels), I think we can conclude that:

(i) With this kind of (consumer line out) signal level the UA-1G peak indicator can indeed be ignored (it seems to be set for -10 dBFS digital output).

(ii) The input level knob can be set to full scale without analog clipping.

(iii) The recorded waveform can be used to confirm that there has been no digital clipping.

(iv) Edit: Noting that the noise floor increases by almost the same amount as the recorded peaks (8-9 dBFS) over this range (2-5 o’clock on the input level knob) the S/N ratio is not going to be improved much (sorry, I got this wrong in the initial post).

The recorded tracks sounded fine to me. Given that plans are afoot to implement 24-foot recording in PartAudio / Audacity for Windows: if you have equipment that can benefit from the lower noise floor, or if you believe in other magical benefits of 24-bit recording (and you don’t pay too much attention to that PEAK LED), the UA-1G may be worth the extra cost. Unfortunately, you may still then need to compile a private version of Audacity with ASIO to get the 24-bit recording capability (or use an ASIO-capable program like Sonar LE which is supplied with the UA-1G device). If you want those extra ins and outs the UA-1G is an obvious choice.

Thank you for the three very informative posts R_G_B.

It may be worth mentioning here that although Audacity does not currently support 24 bit recording on Windows, it does support 24 bit / 32 bit float for importing, editing, processing and exporting. For anyone that wants to take advantage of 24 bit recording, and use Audacity for editing, they could record with an application that supports ASIO (such as Sonar or Wavosaur), then export the recording as either 24 or 32 bit WAV, then import that file into Audacity.

I ought to admit that I could not hear any quality improvement in 24-bit (Sonar / ASIO) over 16-bit (Audacity / DirectSound) recordings from vinyl (without any further application of effects or filters), provided care was taken with levels to use the available dynamic range and resolution without clipping (all with the UA-1G as described above).

In fact with default settings, the 24-bit captures were worse, due to added clicks and pops. I run a lean Win 7 system in which aero and most ‘optional’ startup programs are disabled, and wireless is off when recording. It passed the latency check mentioned below (results around 500 us) but it is still just a netbook with an Atom N450 1.66 GHz processor. Latency is irrelevant to me for transcription from vinyl, so the solution is probably just to increase the buffer size in the ASIO driver.

Anyone who wants to record in 24-bit should listen to some familiar, challenging test tracks recorded in 16- versus 24-bit, and adjust their system settings if necessary before recording in earnest. Recording at 24-bit is more demanding on the computer, and those clicks and pops occur when it can’t keep up, as many people have found to their dismay.

First check your system capability using the free tool at and follow the advice there to locate problem drivers if necessary (and hopefully obtain updates that solve the problem or perhaps disable unnecessary drivers while recording).

Also be prepared to adjust audio buffer settings (in ASIO, I am not sure about WASAPI).

If you still get drop-outs, try turning off background activities such as virus checking (while disconnected from any network for recording sessions) and/or adjust Windows > Performance Options > Advanced to allocate processor resources for best performance of Background Services rather than Programs.

If you record while on battery power, check that your power settings do not reduce processor power while running on battery (or plug into a power brick instead).

Or, if you can’t hear any improvement and don’t want the hassle, stick with 16-bit recording!

Past your running out of computer power at some awkward times, you won’t hear much difference between 16 bit and 24 bit. The reason to use higher bitrates is post production. If you intend to apply multiple filters and tools to the show, errors will appear sooner at lower bit rates and sampling frequencies. It could be argued that 32-floating should be used for everything and downsample to the delivery format.


An argument I support, as it’s what I do to produce WAV files for CD burning and import into iTunes. The CDs when payed on my hihfi ( Rega CD deck, QUAD 33/303 ESL-57 electrostatic speakers) sound extremely good - at least to my aging ears :wink:

I record from my old Technics SL-150 deck with SME 3009 Improved tonearm, through an ART phono pre-amp and then on to an Edirol UA-1EX (the pre-cursor the the UA-1G) set at 16-bit using the latest drivers from the Edirol website.

@R_G_B: may I echo Steve’s thanks for the informative posts - much appreciated.


Recording at 24-bit is more demanding on the computer, and those clicks and pops occur when it can’t keep up, as many people have found to their dismay.

I have successfully recorded 24/192 without any clicks or pop using a Roland EDIROL UA-101 and Audacity running on a Windows 7 IBM Thinkpad.

Yeah once even the screen saver came on but Audacity kept moving.

Just to say it works. And previous to this I have done numerous 24/96 using Audacity on same and TASCAM US 144 … no issues.

@ mpanwar - Indeed so, and even a process running in the background can be the culprit. I got pops in my audio when I had a DVD menu running the background (never thought about it being that tasking). Every few minutes, a nice pop that’s neigh impossible to remove. Grrrrrr…