Pro Audio Files

6 Things to Know About Sample Rate and Bit Depth

In digital audio, sampling rate and bit depth can be a topic of quandary and debate. And with this topic, it can sometimes be difficult to find the facts in a swirling cyclone of opinions, although it is fact that most opinions will remain unchanged. Whether you adopt these facts or not is ultimately up to you. Here are a few things you may not have known about sampling rate and bit depth.

Let’s kick this off by revisiting the basics. We start with an audio wave, a pressure wave with infinite resolution in the time and frequency domain. A wave in which no electronic system — analog or digital — can fully and losslessly capture and reproduce. That being said, the limitations of a digital system is that we only have 1’s and 0’s to represent the audio we are hearing. So we might as well make the best use of those two digits, right?

First, we need to convert the analog representation of our audio wave into the digital realm. For this, we have the analog to digital converter or ADC. Before the ADC, our buttery smooth, and infinitely resolute analog signal must pass through a low pass filter known as an anti-aliasing filter. In short, it’ll prevent frequencies (higher than the ones we’re trying to sample) from interfering with the sampling process. The qualities of these filters play some of the biggest roles in being able to capture an analog sound, and is one of the key factors in what determines a converter’s “sound.”

1. Sampling Rate

Sampling rate is best thought of as a container for frequency information.

It’s the horizontal resolution representing the time domain. It’s arguably not as important as you may think, but does indeed have an effect on the audio you’re trying to reproduce and represent. It should, therefore, still be taken into consideration depending on the delicacy of the recording you are making. Higher frequencies, even those outside our hearing range, do affect the tones that are within that range.

Remember, audio is a pressure wave, so even slight changes in pressure, may induce a butterfly effect with your audio. When we work at higher sampling rates we’re able to capture those inaudible frequencies. These frequencies are going to be the harmonics of the tones that we can hear. But working at higher sample rates also allows us to process them along with the frequencies that we can hear.

So just because we can’t hear those frequencies, doesn’t mean we can’t feel them or that they don’t have an impact on our sound. If we don’t capture frequencies above those in our human hearing, we’d end up with a bunch of high frequency sine waves, without character or subtlety in the treble range.

Often times when we think about sample rate, we’re only thinking of the “highest frequency” we can accurately sample. But we often don’t think about the subtlety of what’s in between. For instance, look at (fig 1a) where we have four 1 sample ticks at a sampling rate of 192 kHz. These ticks represent 1 sample of audio, or ticks that only last 192,000th of a second, which is still audible by the way.

When we use a sample rate converter to resample these same ticks to 44.1 kHz (fig 1b), not only is the frequency information different, but so is the amplitude. So, it’s not only frequency, we’re missing out on dynamic information, too. This happens because of interpolation.

2. Bit Depth

Bit depth is best thought of as a container for the amplitude of our audio.

It’s also arguably not as important in the average home studio with a heavy rock or EDM production. But, if you are listening and recording at proper levels, it can have a large impact during the production process. There’s no sense in cramming the last 4 to 8 bits with all of your audio data, when we have at least 12 to 60 more to work with. With proper gain staging and monitoring levels, these subtleties are much more apparent.

3. Digital to Analog Conversion (DAC)

With both sampling rate and bit depth, the DAC transforms our bits back into a buttery smooth form of analog audio. It’s an interpolated form based on those sampled points. (That is, it fills in the blanks, or guesstimates.) But it’s inarguably based upon the information that was captured. Frequencies outside of the range of hearing also play an important role in the electronics that feed our reproduction equipment. So, it’s important to take that into consideration as well.

Most decent audio equipment is tested up to ranges outside our human hearing and those frequencies do have an impact on the behavior of the electronics in the equipment. Ultimately, we’re trying to digitally capture the analog signal that represents our audio wave. Why spend $3,000 on a preamp if you can’t squeeze every penny out?

It remains true that quad sample rates like 176.4 kHz, 192 kHz and higher may induce sampling errors and other inconsistencies. But in the process of downsampling or decimation later on, we’ll be interpolating and recalculating those errors anyways. The point is to capture the information to begin with. And again, this is arguable to some with regards to its audible effect, but it’s still factual.

4. The Audio File’s Bit Depth

The audio file’s bit depth is often misunderstood and misinterpreted. The audio file that’s on our computer, the same one that is created by our DAW, is simply a container for the information that the ADC already created. So, the data already exists in complete form before it gets into the computer. That’s the key thing to take from this.

When we select an audio file bit depth to record at in the DAW, we’re selecting the size of the container or “bucket” that we want that information to go into. Most ADC’s will be capturing your at audio 24 bit regardless of the selection you make in the DAW.

So what happens when I select a 32-bit or 64-bit float file? Your audio is still 24 bits until it is further processed. In most cases it’ll pass through a plugin effect at 32-bit float, or even a 32-bit mixbus. (Some have 64-bit float.) But with the topic of capturing audio, you aren’t changing anything, or making it sound better by simply putting it into a 32-bit or 64-bit float container. It’s the same information, with just a bunch of 0’s tagged on, waiting for something to do.

So why is a 32-bit or 64-bit float file container good? With a 24-bit file, we have a finite number of (in this case it’s 24) decimal places to capture information between 0 and 1 that our ADC delivers. In a float file, the decimal place can move or “float” to represent different values. Not only that, but we also have an extra 8 bits of resolution or headroom, that wasn’t there before. This allows us to do some pretty impressive things in terms of processing and computing.

We can essentially give our audio more resolution that it originally had, simply by processing it and interpolating new points in the dynamic spectrum. We can also dynamically and non-destructively alter our audio as long as it remains in the digital realm. We can even prevent further clipping of the captured audio. That’s why you hear it’s so important to keep the same bit depth or higher, throughout the entire production process.

So yeah, if you get 16-bit files from your friend to mix down, work at 24-bit, or better yet 32-bit float. You aren’t making it any worse, only better. Whether you are producing subtle classical recordings, or mixing a new breed of square wave EDM, bit depth is just as important at representing your dynamics, even if perceivably there aren’t any.

5. Bit Reduction

This brings us to the important topic of bit reduction. Ultimately, we have to take this audio that is dynamically captured, and make it smaller. So there needs to be yet-again, more interpolation. How do you take 32 bits of information, that you spent so long critically mixing and listening to, and cram it into a 16-bit space? Dithering.

The process of dithering these days doesn’t seem to hold much importance, at least in my small world, and it is often not taken into consideration because of the type of material that’s being produced. With digital audio, the resulting amplitude of an audio signal is a direct representation of its bit depth.

Most popular music now is produced in a way that’s unique, blending distorted tones passing through devices intended to distort them even further (at least in pop music, that is). But it’s part of the style. This distortion creates harmonics, or divisions of frequency, which might be another important argument about why sample rate perhaps should be held in higher regard.

But with loud 16-bit audio, where there isn’t much dynamic movement, and only 16 bits to represent it, we’re only using the last few bits of information to represent our audio. The “sound” of dithering comes when our audio starts reaching the noise floor, but we use that noise floor as a point of comparison for dynamic levels. So make your silence count just as much as your audible information.

6. Downsampling

The topic of choosing a sample rate, and downsampling is also sometimes an exercise in sanity. Here are the facts: 44.1 kHz, 88.2 kHz, and 176.4 kHz are sample rates for audio mediums. Think CDs. If your audio ends up on a CD, that’s what these are for. The sample rates of 48 kHz, 96 kHz, and 192 kHz are for video mediums like DVD’s, blu-rays, etc. The idea of “more is better” is inaccurate. It should be “more is different.” The math to divide 96 kHz to 48 kHz is simple, divide by 2 (compare fig 2a to 2b).


The math to divide 96 kHz to 44.1 kHz (compare fig 2a to 2c) is difficult and therefore changes your sound. It’s not necessarily bad, just different.

The most important thing to understand is that uneven decimation isn’t an accurate representation of what you capture, instead it’s some kind of interpolated form of it. This is why some people believe 48 kHz sounds different from 44.1 kHz. Uneven decimation introduces smearing into the audio (fig 1b) that wasn’t there before. Ringing will always unfortunately be there, more or less.

Another topic of quandary is “should I resample by re-recording the analog signal, or re-sample in the computer?” Both methods certainly have their pros and cons. By re-recording the analog playback of a digital recording, we are simply re-capturing an interpolated waveform which lends itself to greater analog accuracy. However, it is possible to introduce unwanted noise, so it’s something to watch out for.

If you already have an excellent signal-to-noise ratio, it isn’t necessarily introducing anything that’s noticeable in the confines of our bit depth, and would certainly be taken care of after dithering. By resampling “in the box,” we can end up with those ringing artifacts that you see in the different pictures. It’s an unavoidable scenario unfortunately as a result of math. Some algorithms do a better or “different” job than others. In the end, it truly is up to your ears. Still, it’s important to know the facts. So with that, I’ll get off my high horse and we’ll see you next time!

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Dave Askew

Dave Askew

Dave is a tech, trainer, and consultant specializing in studio integration and audio production. He's traveled coast to coast and helped some of the most respected names in the industry. Find his tutorials at Groove3 and connect with him at LinkedIn or contact him at Media | DMA.

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  • Nelson T. Gast

    So I see the advantages of the higher rates and whatnot from a standpoint of measuring audio, but my question for you, Dave, is: can/could you (you, not one) hear the difference in a blind test?

    • Hi Nelson! Thanks for sharing, fair question. If you need to know for curiosities sake, then yes I can but only in an A/B environment. As mentioned in the other linked articles, subjectivity remains a matter of taste. Just presenting the facts here. Let me expand on this idea though just to clarify a few concepts that inspired the article. Keep in mind I’m not a proponent or opponent of recording at a given sample rate. In the end you are the creator and make the decision.

      In a critical listening environment it’s not too difficult to hear the difference, if you know what to listen for. However, the difference is NOT like opening a door to another dimension. The idea of oversampling is in the spirit of preservation. And of course, your sound will only be as good as it’s weakest component. Would I be able to sit down and tell if someone originally captured audio and put it on a CD at 88.2 or 44.1? No. You surprise yourself with the differences you hear when doing A/B comparisons and what your ear will pick up on, ESPECIALLY when analog equipment comes into play. The premise is to maintain signal detail and integrity. There is ONE possible caveat to this, which I’ll talk about in a little bit.

      First, you’ll need to determine if you like the effect of the anti-alias filter on the ADC for given sample rate, then go from there. Another idea is that using higher sample rates may even reduce the effect of the anti-alias filter on the frequencies you are trying to capture simply by pushing it up out of the way. You can then rely on something else further down the line during the conversion process to do the AA and conversion, instead of relying on an ADC with a budget AA Filter. There is a decent Wikipedia article on anti-aliasing that may further strengthen the case of those that use oversampling. PSP Audioware uses this unique concept of oversampling mixed with truncating in their Master Q2 plugin. The idea is to push the ugly stuff out of ears reach, then chop it off.

      The qualitative aspects of the subject matter likely stray from a typical home studio setup, the audible difference ultimately comes down to the quality of gear that recreates the sound, and how well your acoustic space is set up. I wouldn’t “hear” the subtlety sitting on the john with my iPod in an NYC subway bathroom while doing A/B comparisons. I wouldn’t even necessarily hear the difference with substandard equipment. The best way I can describe it is that there is an “openness and transparency” to the sound. Does the end user care? Not likely. Trained ears hear detail though when presented to them in a controlled environment.

      I’m not talking about “hearing ultra sonic frequencies”, for example I can’t hear dog frequencies, no one can. The effect that those inaudible frequencies have on the captured waveform we intend to manipulate, however, is present. Particularly while is being processed in the digital domain. Any time we combine digital and analog equipment, frequency ranges much higher than what we work with in the digital domain will always be present, and be manipulated. Here is the exception I was talking about previously. The sample jockey or “samplist” to keep it pc, who uses pre-recorded beats or VST instruments and samples, keeping everything inside the box will unlikely hear much difference at all, due to the fact it is working with pre-recorded material and being processed in a digital domain. In the spirit audio engineering, making things sound as good as possible, and preserving the art that is being captured, it only makes sense to not only capture the audible sound, but also preserve the environmental subtlety (for lack of better terms).

      Here is an example that may help define what I’m speaking of. Have you ever noticed in high action sequences in a movie, that effect where there is no motion blur and everything is sharp and crisp? It adds some intense clarity (Saving Private Ryan comes to mind). They achieve that by shooting at a higher framerate or shutter speed (sample rate in our case), then re-sampling it later at the desired frame rate that is of an equal division. There is more to it than that in the video realm, but that’s the idea. This reduces the motion blur between frames, which at this point represents interpolation between audio samples samples. So not only will you get a more accurate interpolation, but a more accurate sample in terms of positioning, particularly when it relates to bit depth.

      The intent of the article was to keep it factual though and stay away from the subjectivity of the topic, because these types of topics can easily go in circles. We can’t argue the facts though. The main point is to use and understand these facts, experiment, and trust your own ears.


    I’m still trying to understand digitizing audio signals. But from what I understand the digital information that is stored in an audio file isn’t actually information about an audio sound. What it is is the digital information about a modulated carrier signal, which in turn is information about an audio sound… I think??

  • Joe Smith

    Bottom line folks, and what isn’t given here: recording above 48khz (heck above 44.1 realistically) not only buys you nothing, but if anything, will make your recordings slightly worse..and ridiculously
    bigger in file size. Don’t buy the “more is better” silliness.

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