Audio Recording

Most recording gears or software defaults to 48 KHz as the sampling rate during digital recording. For example; Ardour an open source DAW defaults to 48 KHz. Other than this; there are other DAW and recording hardware that defaults to 48 KHz.

On the opposition, lots of folks practically the recording industry professionals (engineers, mixers and mastering engineers) prefer to use 44.1 KHz as the sample rate during digital audio recording and not 48 KHz. And there are a lot of recording software and DAW that defaults to 44.1 KHz as the sampling rate that includes Reaper.

So the main question is: Which is better 44.1 KHz or 48 KHz when recording audio in digital?

48 KHz is too much when recording audio?

The Nyquist theory states that to accurately reproduce analog audio in digital it should be sampled at least twice the maximum audible frequency. This is where confusion starts to sets in and debates in recording forums are too common. In schools, you learned that human ear can only perceived 20Hz to 22,000 Hz. Human adults have lower treble hearing response and lies around 15,000 to 16,000 Hz. Doubtful with this, I do a hearing test from 20Hz to 20,000 Hz and was surprised I could not anymore hear frequencies above 16,000 Hz.

If the maximum audio frequency is 22,000 Hz by theory, the sampling rate required is:

Sampling rate = 2 x Maximum Audio Frequency = 2 x 22,000 Hz ~ 44.1 KHz

This is where 44.1 KHz came and why it is commonly used as the sampling rate for digital audio applications such as in recording and in digital audio playback.

If the sampling rate is 48KHz, reverse calculation reveals that the maximum audio frequency that can be sampled is:

Maximum Audio Frequency = Sampling Rate/2 = 48KHz/2 = 24 KHz

If Human Adults or even any humans cannot hear above 22,000Hz why other engineers or recording gears still stick to 24 KHz as the audio sample rate?

At first, you would realize that 48 KHz sampling rate is too much for recording digital audio because you are recording frequencies that could not be heard anyway. But for perfect reproduction, it takes more than to simply hear these frequencies.
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This is one of the common questions in home recording even today. Before answering directly this question, you need to know some important history on how digital recording process evolved over time. Back in the very old days (when USB and Firewire audio interface are not yet common in the market), it is often very difficult to get a good recording signal without a mixer. It is because the first type of audio interface used is a PCI sound card which does not have the sufficient electronic circuitry to get clean recorded signal.

Most PCI sound cards when used for professional music production have some serious limitations which are as follows:

a.) The recorded signal is often weak and noisy because there is no quality microphone pre-amp in any PCI sound cards.

b.) Most sound cards available during that time accept at most two channel recording (stereo) at the same time. If you are tracking a band which requires more than two channels, it is impossible to do that with a PCI sound card alone.

c.) The impedance mismatch between the sound card line input and musical instruments output are severe, and you will notice a weak and noisy signal as a result.

Without a mixer, the most primitive connections of musical instrument to your computer PCI soundcard are as follows:

Musical Instrument — PCI Soundcard – Computer (DAW)

In terms of signal path and recording, you can only record one instrument at a time because of input and sound card limitations:

Home Studio Mixer is the immediate answer before

Plague with so many recording quality issues, the problems are solved by using a home studio mixer. The required mixer may not need to be very big or expensive. Even a small Behringer Xenyx 502 would work. Thus it is connected as follows:

Musical Instrument — Home Studio Mixer – PCI Soundcard – Computer (DAW)

Now the musical instruments (guitars, microphones, etc) are connected first to a small mixer before connecting to the PCI soundcard inputs. The mixer has a built in preamp that boost the microphone levels to line level outputs. With this setup, it finally solves the recording signal quality issues associated with the original PCI soundcard recording before.
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If Shure SM58 is a legendary dynamic microphone, it is OK to use them for recording vocals in my home studio?

The immediate answer is YES. But why you can use them deserves an entire blog post. There is more to learn than by simply buying SM58 and call it a perfect vocal recording solution. First it is important to know its frequency response characteristics. See below:

Photo credits: Shure.com

This is the behavior of SM58:

1.) It is linearly/flat somewhere around 110Hz to 2500Hz. This implies that if the original vocals lie within this frequency range, it is neither boosted nor cut by the microphone behavior. The microphone does not add any EQ coloration at this frequency range.

2.) But if the vocal frequency is somewhat below 110Hz, the vocals frequency would be roll off (cut or attenuated) even if there is substantial bass presence in the vocals.

3.) Now for frequencies above 2500Hz until 10,000Hz, SM58 would gradually add a boost as shown in the frequency response peaking at +5dB somewhere between 5000Hz and 10,000Hz.

4.) It adds a very slight boost above 10,000Hz but not significant above 15,000Hz.

What does SM58 Frequency Response Implies in terms of Recording?

Since SM58 is a perfect vocal microphone for live performance, the frequency response as well confirms it can be a great microphone for recording vocals in your home studio in such a way that it absolutely needs less processing in the mix resulting in a more natural vocal recording. The following are the reasons why:
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