Digital Audio Basics: Sampling Rate and Bit Depth

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Digital Audio Basics: Sampling Rate and Bit Depth

How does the sampling rate affect digital audio quality?

When it comes to digital audio, the sampling rate plays a crucial role in determining the quality of the sound. The sampling rate refers to the number of samples taken per second to represent the analog audio signal digitally. A higher sampling rate captures more details and provides a more accurate representation of the original sound wave.

Let’s take the example of CD-quality audio, which has a standard sampling rate of 44.1 kHz. This means that 44,100 samples are taken every second to record the audio. With a higher sampling rate, a wider frequency range can be captured, resulting in better audio fidelity and a more immersive listening experience. However, it’s important to note that increasing the sampling rate also requires more storage space.

Benefits of higher sampling rates

Higher sampling rates, such as 96 kHz or 192 kHz, offer several benefits for audio enthusiasts and professionals. Firstly, they allow for more accurate representation of high-frequency sounds, resulting in improved clarity and detail in the audio. Secondly, a higher sampling rate provides greater headroom for digital processing and effects, allowing for more precise manipulation of the sound.

Factors to consider when choosing a sampling rate

While higher sampling rates offer improved audio quality, there are a few factors to consider when choosing the appropriate sampling rate for your needs. Firstly, the capabilities of your audio playback devices and equipment should be taken into account. Not all devices support high sampling rates, and using a higher sampling rate without compatible hardware may not yield the desired benefits.

Additionally, the nature of the audio content should be considered. For example, music production and professional audio mastering often make use of higher sampling rates to capture the intricacies of the sound. On the other hand, for casual listening or streaming, a standard sampling rate like 44.1 kHz may suffice.

Understanding the significance of bit depth in digital audio

Bit depth is another crucial aspect of digital audio that directly impacts the dynamic range and accuracy of the sound reproduction. It refers to the number of bits used to represent each sample in a digital audio file. A higher bit depth allows for more precise quantization and results in a greater dynamic range.

Commonly used bit depths in audio production include 16-bit and 24-bit. A 16-bit audio file can represent 65,536 discrete volume levels, while a 24-bit audio file can represent a staggering 16,777,216 levels. This greater dynamic range enables the reproduction of softer and louder sounds with more accuracy, reducing the potential for quantization noise and distortion.

The impact of bit depth on audio resolution

Bit depth directly affects the resolution of the audio signal. A higher bit depth provides finer resolution and allows for more subtle details to be captured in the recording. This is particularly important when working with audio that has a wide dynamic range, such as classical music or film soundtracks.

Choosing the appropriate bit depth

The choice of bit depth depends on the nature of the audio content and the intended usage. For most consumer applications, a 16-bit depth is sufficient to deliver high-quality audio. However, for professional audio production or situations that demand the utmost accuracy, a 24-bit depth is recommended.

It’s worth noting that higher bit depths also result in larger file sizes, as more data is required to represent each sample. Therefore, considerations such as available storage space and transmission bandwidth should be taken into account when selecting the appropriate bit depth.

Final Words

Understanding the basics of digital audio, including sampling rate and bit depth, is essential for anyone seeking to optimize their audio experience. By choosing the right sampling rate and bit depth, you can ensure that your digital audio files accurately capture and reproduce the nuances of the original sound, whether you are an audio professional or an avid music lover.

Remember, when it comes to sampling rate, higher rates offer increased fidelity and clarity, but it’s essential to consider compatibility and storage requirements. Similarly, higher bit depths provide greater dynamic range and resolution, but the choice should be based on the nature of the audio content and intended use.

By mastering the fundamentals of digital audio, you can unlock a world of rich and immersive sound, enhancing your enjoyment of music, movies, and other multimedia experiences.

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What Is Audio Sampling Rate: A Comprehensive Explanation

Introduction

Audio sampling rate is a fundamental concept in digital audio that refers to the number of samples per second used to represent an analog audio signal in digital form. In this article, we’ll explore the technical details of audio sampling rate, its importance in digital audio, and its impact on audio quality and file size.

Sampling Rate Fundamentals

The concept of audio sampling rate is based on the Nyquist-Shannon sampling theorem, which states that in order to accurately represent an analog signal in digital form, the sampling rate must be at least twice the highest frequency present in the signal. This means that a signal with a highest frequency of 20kHz (the upper limit of human hearing) must be sampled at a rate of at least 40kHz in order to be accurately represented.

Sampling rate is measured in Hertz (Hz), which refers to the number of samples per second. Common sampling rates in digital audio range from 44.1kHz (used in CDs) to 192kHz (used in some high-resolution audio formats).

Sample Rate Conversion

In some cases, it may be necessary to convert audio from one sampling rate to another. Sample rate conversion involves resampling the audio data to a different rate, which can be done using digital signal processing techniques. However, sample rate conversion can introduce artifacts and reduce audio quality, especially when downsampling from a higher rate to a lower rate.

There are various reasons why sample rate conversion may be necessary, such as when mixing audio tracks with different sampling rates, or when preparing audio for distribution on different platforms with varying requirements.

Audio Quality and Sampling Rate

The sampling rate has a significant impact on audio quality, with higher sampling rates generally resulting in better fidelity and more accurate representation of the original signal. However, the benefits of higher sampling rates are limited by the limitations of human hearing and the practical limitations of digital audio technology.

While there is debate about the benefits of “high-resolution audio” formats with sampling rates above 44.1kHz, it is generally accepted that sampling rates above 96kHz provide little additional benefit in terms of audio quality.

Bit Depth and Sampling Rate

The bit depth of an audio sample refers to the number of bits used to represent the amplitude of the signal at each sample point. Higher bit depths allow for more precise representation of the signal, but also result in larger file sizes. The bit depth and sampling rate are related, as increasing the bit depth requires more data to be stored for each sample.

There is a trade-off between sampling rate and bit depth, as higher sampling rates require more data to be stored per second, which can limit the maximum bit depth that can be used without exceeding practical file size limits. However, this trade-off can be mitigated by using efficient audio compression techniques.

Sample Rate in Practice

Common sampling rates in digital audio include 44.1kHz (used in CDs), 48kHz (used in digital video), 88.2kHz, 96kHz, 176.4kHz, and 192kHz. Streaming services such as Spotify and Apple Music typically use lower sampling rates for their audio streams, with 44.1kHz being a common choice.

The Nyquist Theorem, named after the Swedish-American physicist Harry Nyquist, states that the sampling rate should be at least twice the highest frequency component in the signal being sampled. This is why the standard CD quality sampling rate is 44.1 kHz, which is just above the upper limit of human hearing.

However, it is important to note that there are higher sampling rates available, such as 48 kHz, 96 kHz, and even 192 kHz. These higher sampling rates can provide more detail and accuracy in the digital representation of the analog signal. However, they also require more storage space and processing power.

Another important factor to consider is the bit depth, which is the number of bits used to represent each sample. The more bits used, the more accurate and detailed the representation of the analog signal. CD quality uses a bit depth of 16 bits, but higher bit depths such as 24 bits are also available.

It is worth noting that some argue that higher sampling rates and bit depths may not necessarily result in audible improvements in sound quality, especially when considering the limitations of human hearing. Additionally, some argue that the increased storage and processing requirements may not be worth the potential improvements.

In conclusion, the sampling rate is a crucial component in the digital representation of analog audio signals. A higher sampling rate can provide more detail and accuracy in the digital representation, but also requires more storage and processing power. The Nyquist Theorem provides a guideline for choosing the appropriate sampling rate based on the highest frequency component in the signal. Additionally, the bit depth is another factor to consider in the accuracy and detail of the digital representation. While higher sampling rates and bit depths are available, the potential improvements in sound quality must be balanced against the increased storage and processing requirements.

Bitrate Part 2

bitrate

The amount of information transmitted through the channel per unit of time is called the bit rate, and the unit is bits per second (bit/s), called the bit rate.

BITRATE

Bitrate is often used in communications as a synonym for connection speed, transmission speed, channel capacity, peak throughput, and digital bandwidth capacity. The higher the bit rate, the higher the data transfer. Bit rate in video refers to the sampling rate at which an analog signal is converted to a digital signal [4] . Video file quality is often measured in terms of bitrate. [4] .
Distinction of conceptedit transmission
Baud rate is also known as waveform rate or modulation rate. The code for a data unit is represented by a finite combination of numbers, each of which is a symbol (or code point). In electrical communication, an electrical waveform is often used to represent one or more symbols. Waveforms with different characteristics may represent different symbol values or symbol combination values, and the duration of the waveform corresponds to the duration of the symbol or symbol combination it represents. Obviously, the shorter the duration of an electrical waveform, the more waveforms are transmitted in a unit of time, or the more data is transmitted, that is, the higher the data rate. Therefore, we can define the baud rate as follows: In the process of data transmission, the number of waveforms transmitted per unit time on the line is the baud rate, and its unit is “baud” [5] .
“Bit rate” and “baud rate” are speed units defined in two different concepts, and it is often easy to confuse them when you are not careful. When binary waveform is used, baud rate and bit rate have the same value, but their meanings are different [5] .
Difference: Both bit rate and baud rate are units that measure the transmission rate of a modem. In data transmission, data information is represented by binary numbers “0” and “1”, and each binary number is called 1 bit. The number of bits transmitted through the channel per unit of time is called the bit rate, expressed in bits per second, usually abbreviated as bit/s. The number of symbols transmitted through the channel per unit of time is called the baud rate, also called the modulation rate. Bit rate and baud rate are consistent only when modulated with two values. For example, in quadrature modulation, every two bits of the data signal form a symbol, and there are 4 values: 00, 01, 10 and 11, which represent the phase changes of the 4 types of carrier signals respectively, for Therefore, send such a symbol. It is equivalent to transmitting two bits of data, and the baud rate is equivalent to half the bit rate. The usual transmission rates of 300, 600, 1200 and 9600, etc., refer to the baud rate, which indicates that the number of binary numbers transmitted per unit of time is 300, 600, 1200 and 9600 [6] .

Bit rate

Bit rate refers to the number of bits (bit) transmitted per unit of time, in bps (bit per second).

Bit rate is also known as “binary bit rate”, commonly known as “code rate”. Indicates the number of bits transmitted per unit of time. It is used to measure the transmission speed of digital information, often written as bit/sec. According to the number of bits occupied by each image storage frame and the transmission bit rate, the digital image information transmission speed can be calculated [1].
In modern digital communication, the transmission volume of digitized video and other information is large, so it is often measured in kilobits per second or megabits per second, which are written as kbit/sec (or kbps) and Mbit/sec. (or Mbps respectively). ). For example, the amount of information digitized from an ordinary color TV signal can reach 216 Mbit/sec. A good digital broadcast channel can transmit dozens of color TV programs, and its capacity can reach several gigabits or gigabits per second (written as Gbit/sec or Gbps) [1] .
Bitrate is often used to measure the quality of video files.
Bitrate is often used to measure the quality of video files.
flexibility edit stream
Because each network is unique and each access line has different conditions (such as length, attenuation, crosstalk environment, etc.), access lines from different telephone companies must support different data rates. For ADSL and VDSL modems, it is best to set the data rate to one of many possible data rates. For example, DMT-based ADSL and VDSL can theoretically change the tariff at fine intervals, and CAP-based RADSL (Rate Adaptive ADSL) also provides some flexibility in tariff configuration [2].
However, telephone companies may want to limit xDSL service to a small set of rates sufficient to provide a variety of services. If a limited set of tariffs can be adapted to a wide range of services, then the management of the services in this case is simpler than in the case of variable tariffs. Telephone companies want the choice of modem speed to be under the control of the network, not the user [2] .
In this mode, the selection of the transmission rate set of the xDSL network must be prudent. In this case, there is a possibility that two adjacent systems receive traffic at very different rates and the system must be able to handle such a situation. The other model, the “best match” approach using adaptive rate ADSL (similar to a voiceband modem), is more beneficial to new network operators and Internet Service Providers (ISPs) [2] .
Transmission control method
Most bit rate control schemes consist of two parts. Part of the encoded bit stream output by the encoder is fed into a buffer. For a constant bitrate channel, the data in the buffer is fetched at a constant rate, and if the buffer is large enough, the bitrate variation caused by the MPEG picture type, etc. can be smoothed out. This is necessary for both constant bit rate transmission and variable bit rate transmission in general. However, in practice, the buffer size is always limited. The buffering process will bring a delay to the system, and this delay is proportional to the size of the buffer. Latency is often a serious issue for real-time image communication, so buffers should be kept as small as possible. That is, long-term fluctuations in bitrate due to changes in scene content or changes, etc. they cannot be softened in this way, so another part is needed. This is to send some measure of the output bitrate to the encoder to control the encoding process, thus changing the output bitrate [3] .

Sample rate and bit rate of MP3 Part 2

BIT RATE

The number of digits in the sound is equivalent to the number of colors on the screen, indicating the amount of data per sample.

Of course, the larger the amount of data, the more accurate the playback sound, so as not to confuse the sound. of the teapot with the train whistle. In the same way, it is more clear and precise for the image, so as not to confuse blood and ketchup. [However, limited by the function of human organs, 16-bit sound and 24-bit image are basically the limits of ordinary humans, and the higher digits can only be distinguished by instruments. For example, the phone has 7-bit sound sampled at 3 kHz and the CD has 16-bit sound sampled at 44.1 kHz, so the CD is clearer than the phone. ]

When you understand the above two concepts, bitrate is easy to understand. Take the phone as an example, 3000 samples per second, each sample is 7 bits, then the phone’s bit rate is 21000. And the CD is 44100 samples per second, two channels, each sample is 13 bit PCM encoded, so the CD bit rate is 44100*2*13=1146600, which means the CD data volume per second is about 144KB. the capacity of a CD is 74 minutes equal to 4440 seconds, which is 639360KB=640MB.

Sound is actually a type of energy wave, so it also has the characteristics of frequency and amplitude, with frequency corresponding to the time axis and amplitude corresponding to the level axis. The wave is infinitely smooth, and the string can be considered to be made up of innumerable points. Since the storage space is relatively limited, in the process of digital encoding, the points of the string must be sampled. The sampling process consists of extracting the frequency value of a certain point. Obviously, the more points that are extracted in one second, the richer the frequency information that can be obtained. To restore the waveform, there must be two sampling points in one vibration. The highest frequency that can be felt is 20kHz, so to meet the auditory requirements of the human ear, at least 40k samples per second, expressed at 40kHz, and this 40kHz is the sample rate. Our common CD has a sample rate of 44.1 kHz. It is not enough to have only frequency information, we must also obtain and quantify the energy value of this frequency to represent the strength of the signal. The number of quantization levels is an integer power of 2, and the sample size of our common CD bit is 16 bits, that is, 2 to the power of 16. Sample size is harder to understand than bit rate. sampling, because it makes it seem abstract. For a simple example: suppose a wave is sampled 8 times, and the energy values corresponding to the sampling points are A1-A8, but we only use 2-bit sampling size, as a result we can only keep the 4 point values in A1-A8 and discard the other 4. If we use the 3bit sample size, all 8 point information is recorded. The higher the sample rate and sample size values, the closer the recorded waveform is to the original signal.

MP3 sample rate and bit rate

Bit Rate

When we listen to mp3 and watch movies, we will notice two parameters.

BIT RATE

The most common ones are 44.1 KHz sample rate and 192 Kbps bit rate. So what is the sample rate and what is the bit rate? What is the relationship between them? Explain:

The process of converting an analog audio signal to a digital audio signal is called sampling. In a nutshell, how many data points does it take to record a 1 second long sound via waveform sampling. For example: the sound sample rate of 44.1 KHz is equivalent to spending 44,000 data points to describe the sound waveform for 1 second. In principle, the higher the sample rate, the better the sound quality; sampling frequency is generally divided into three levels: 22.05KHz, 44.1KHz and 48KHz; 22.05KHz can only achieve FM radio sound quality, and 44.1KHz is the theoretical limit of CD sound quality, 48KHz has reached DVD quality.

Sampling rate refers to the sampling frequency when converting sound (analog signal) to mp3 (digital signal), i.e. how many data points are sampled per unit of time. (The data for a sample point is 8 (or even more) bits long.)

Bit rate refers to the number of bits (bits) transmitted per second. The unit is bps (bit per second). The higher the bitrate, the more data transmitted and the better the sound quality.

It can be said that the sample rate and bit rate are like the horizontal and vertical coordinates on the coordinate axis. The sampling frequency on the abscissa represents the data points sampled per second. The bit rate on the ordinate represents the precision when quantizing analog quantities with digital quantities.

The sample rate is similar to the number of frames of moving images. For example, the sampling rate of movies is 24 Hz, the sampling rate of PAL format is 25 Hz, and the sampling rate of NTSC format is 30 Hz. When we play back the still images sampled at the same rate as the sampling frequency, we see a continuous image. In the same way, when a CD recorded at a sampling rate of 44.1 kHz is played back at the same rate, a continuous sound can be heard. Obviously, the higher the sample rate, the more coherent the sound will be heard and the picture will be seen. [Of course, the sampling rate that human auditory and visual organs can distinguish is limited, which is basically higher than sound sampled at 44.1kHZ, and most people haven’t noticed the difference. ]

Quality (bit rate)

In multimedia technology, quality is often used to judge the effect of audio, and quality here is actually bitrate.

1. Introduction
2 sound control
3 encoding mode
Introductionedit transmission
The term quality is widely used.
In multimedia technology, quality is often used to judge the effect of audio, and quality here is actually bitrate.
On WINDOWS it is called “bit rate” and on some players it is described as ” bit rate “.
Quality refers to the bit rate at which digital sound is converted from analog to digital format. The higher the bitrate, the better the quality of the restored sound.
sound control edit stream
16 Kbps = phone quality
24 Kbps = increase phone quality, shortwave transmission, longwave transmission, European standard medium wave transmission
40 Kbps = American standard medium wave transmission
56Kbps=Voice
64 Kbps = boost voice (best bitrate setting for cell phone ringtones, best setting for cell phone mono MP3 players)
112 Kbps = FM stereo broadcast FM 128 Kbps = tape (best setting for mobile phone stereo MP3 player, best setting for low-end MP3 player)
160 Kbps = HIFI high fidelity (best setting for mid to high end MP3 players)
192Kbps=CD (best setting for high-end MP3 players)
256Kbps=Studio Music Studio (for music enthusiasts)
In fact, with the advancement of technology, the quality of music is also getting higher and higher, the highest quality of MP3 is 320Kbps, but some formats can achieve higher sound quality.
For example, the emerging APE audio format can provide real audiophile level lossless sound quality and smaller volume than WAV format, and its quality is usually 550kbps-950kbps.
encoding modeedit stream
VBR (Variable Bitrate) Dynamic Bitrate means there is no fixed bitrate. The compression software immediately determines which bitrate to use based on the audio data being compressed. This is a method that takes quality as a premise and takes file size into account The recommended encoding mode;
ABR Average Bit Rate (Average Bit Rate) is an interpolation parameter of VBR. LAME created this encoding mode in response to the low file volume ratio of CBR and the variable size of files generated by VBR. Within the specified file size, ABR takes every 50 frames (about 1 second for 30 frames) as a segment. High-frequency and insensitive frequencies use relatively low traffic, and low-frequency and large dynamic performance use high traffic, which can be used as VBR and CBR, a compromise option.
CBR (constant bitrate), constant bitrate means the file has one bitrate from start to finish. Compared to VBR and ABR, the compressed file size is very large and the sound quality will not improve significantly compared to VBR and ABR.

Some details of the sample rate

For many years it was thought that the sample rate or sampling frequency did not decisively influence the final quality of the digital audio; There are currently several engineers who record in 44.1K or 48K without really knowing why they do it. With the advent of new and better computers, interfaces, ports and protocols, 88.2K, 96K and up to 192K entered the discussion table on the best sample rate to use. It has always been the subject of discussion between engineers and audiophiles; some argued that they did hear the difference between different sample rates and others that did not, and the topic has been subjected to millions of A / B tests with very high quality equipment, causing all kinds of opinions found and uncompromising, fights and friendships of years broken

While this is a basic issue of digital audio, it is always surrounded by a halo of mystery, mysticism and magic (like every sound theme), which is well worth clarifying.

What is the sample rate?

This topic, although it occurs in the first or second class of digital audio, is not always understood correctly. In scholastic thinking, sample rate is defined as the amount of audio samples transported and taken per second. Since this is a unit of measurement over a second and with events that occur cyclically, the Hertz (1 / Frequency) is used as a unit. Obviously we cannot talk about this subject without referring to the Nyquist sampling theorem, which was tested by Shannon almost twenty years after its publication and in which it is stated that for a signal of limited bandwidth (B) (for example, a vibraphone reaches 14.917Hz), the sampling frequency must be twice its bandwidth (2 * B). Then, taking the previous example, we can say that: 2 * B → 2 * 14.917Hz → The sampling frequency for 14.917Hz should be 29.834Hz. This would be equivalent to 29,834 samples per second (1/29, 834) to be able to regenerate the signal of a vibraphone without error. Hence, it is taken that the highest frequency that human beings listen to is 20kHz and if we apply Nyquist it should be 40kHz, but it takes 44.1kHz to meet the demanding ears and for a matter of multiples.

44.1K or 48K to 88.2K or 96K, the correct division

At the dawn of the digital audio era, Nyquist was used to use the sampling resolution of 44.1K, used at that time audio CD format that played at 16bit / 44.1kHz. With the advent of DVD and Blu Ray as video and audio formats, resolutions such as 24Bits / 48K or 24Bits / 96kHz began to be used. Although for many years there were recordings that were made in 24Bits / 88.2kHz or 24Bits / 96kHz, at a certain time of mastering, before sending it to the disk duplicator, the audio suffered a mutilation that reduced it to 16Bits / 44.1kHz as It was ordered by the CD format. This process should be carried out with equipment specially designed for this function and in stages so that the audio did not suffer a very noticeable cut and the bad conversion was evidenced. Although the old and dear Dither was applied since then to compensate for this process (something like “grain” in the cinema. Watch a film without “grain” and it will look like HD even though it was filmed in 1980 on tape and goes to notice until the makeup of the actor and the assembly of the special effects, something otherwise disagreeable).

Generally, to prevent the audio from mutilating or applying several conversions that degrade it, it was decided at what resolution to record before pressing the REC button (we will not mention those that come down directly with your DAW from 24Bits / 96kHz to 16Bits / 44.1kHz in one step to export the audio … there is a place reserved especially for them in hell). If the audio was going to end on CD, a 88.2kHz sample rate was generally applied, since at the time of mastering, with the symmetric re-sampling at “half”, it was 44.1kHz.

Sounds better?

The subjective point of this is that we expect recordings to “sound” better at a higher sample rate. The reality is that if we record in high sample rates, with very good sampling, our sound will not “sound better”, but will be more detailed. Obviously, if our sound source is bad, our microphones and preamps too and so on, no matter how much we record at 192K, the result will not be the best. Now, if we use a good sound source, good audio chain and a good converter, everything will be obviously good. But don’t confuse; We are talking about detail here, not if it will sound more “warm,” “fat,” or “full-bodied.” This translates into a more homogeneous capture of the entire frequency spectrum, both audible and non-audible.

CPU, disk and plug-ins

Obviously, having a higher sample rate means that our processor must do more calculations, since it has to process more samples (or audio samples). Depending on the amount of plug-ins that we use before a multitrack in high resolution, our use of both DSP and native processors (the computer equipment), will increase significantly, making it very difficult or impossible to work. There are several options to overcome this problem, from buying more processor or DSP, using fewer processes or external equipment (hybrid mixing), to borrowing a machine. The only option that should never go through our minds is to lower the resolution of the audio, process and upload it again. The serious problem that comes with this is a cut in the audio, which is not reversible and what is limited and trimmed, so it stays.

Another aspect to consider is that the storage speed must be in accordance with the audio resolution we use. Suppose we want to record at 24Bits / 96kHz; The transfer rate would be: 2304kbits / second. Now, calculating the amount of tracks, we should use a disc that really reaches us in speed for this transfer rate (topic to be developed in another article).

In these times, storage size is not a problem, but speed is. Having three terabyte disk drives are generally used for 5400 rpm dish disks; the least that should be used if they are not solid state disks, would be 7200 rpm plate disc drives. Obviously, with 5400 rpm discs, we would have a third reduction in the final transfer speed and reading and writing possibilities called “iops” (in out per second or in and out per second), which have a certain number, depending on the disk, capacity and arrangement of the same (RAID) which, depending on how much we demand in the resolution of the audio, amount of channels, processing (plug-ins) and expected latency (if we record with real-time monitoring), we will surely face some problems like “clicks” and / or “pops” in our audio.

Clock

The importance of using a good clock (or clock) and being in sync with all the elements that belong to our audio chain is vital. Recall that a few articles ago we have exposed this topic in detail, but it should be reinforced in this article. Several ADC and DAC converters of economic interfaces do not perform sampling and quantization in the correct or expected manner; External clocks or protocols such as Dante help the synchronization between several devices to be correct and improve the audio quality. Much of the final quality of our work in audio is in this part of the process and it is important that if we take our work and passion seriously, we begin to pay attention to these kinds of details that are generally overlooked.