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MP3 bitrate encoding mode

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MP3 bitrate encoding mode

MP3 bitrate

Bit rate of 1 MP3
Generally, there are three mp3 bitrates namely VBR, ABR and CBR.

mp3 bitrate

1.1 RBC
CBR is short for Constant Bit Rate, which means Fixed Bit Rate in Chinese.

For a CBR MP3 song with a bitrate of 128kbps, the first 128kb of the song describes the sound of the first second, and the second 128kb describes the sound of the second second…if the song is finished, it will take 640 seconds, then the song size is 128kb × 640 = 80Mb = 10MB. The so-called 128kbps means 128kb per second.

If you are careful, you will find that the volume compressed by this encoding method will be very large, because the bit rate is fixed. Of course, the sound quality has some advantages over the other two, although this advantage may be minimal.

1.2VBR
Dynamic bit rate VBR (Variable Bitrate). That is, there is no fixed bitrate and the compression software determines on the fly which bitrate to use based on the audio data being compressed.

A simple understanding is that the bitrate will be relatively high at the time the song is rich in detail, and relatively low at other times, so sound quality and size are taken into account. For example: at the beginning of the song, a person sings alone, the sound is relatively simple, we use 64kb to describe the sound within one second; at the climax of the song, everyone sings, the sound is more complicated, we use 256kb to describe a second voice within the species.

1.3 APR
ABR (Average Bit Rate) Average Bit Rate is an interpolation parameter of VBR.

For example, when you specify 192kbps ABR to encode a wav file, Lame will use a fixed 192kbps encoding for 85% of the file, then dynamically optimize the remaining 15%: complex parts are encoded with more than 192kbps, simple parts are encoded with less than 192 kbps. Compared to CBR 192kbps, ABR 192kbps has a similar file size, but the sound quality is much better. ABR encoding is 2 to 3 times faster than VBR encoding and has better quality than CBR in the range of 128 to 256 kbps.

It can be used as a compromise between VBR and CBR. Under normal circumstances, files with this encoding method are rarely found.

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The relationship between MP3 frequency, bit rate, bit rate and sound quality. Part 2

MP3 ENCODING

Speaking of mp3, I am afraid no one will say that they have never heard of it.

mp3 encoding

Even if you are not an mp3 user, there are ubiquitous advertisements, advertising activities in the city, discussions between friends and the Internet. Rich resources, these always give you a little impression, right? For trendy youngsters, especially friends who like music and friends who like digital devices, mp3 is probably a word that should be talked about every day, but what is mp3, how to determine mp3 sound quality and what is good or How can I listen to high quality mp3? ? ? I think the following article can help you solve many doubts.
Across current mp3 users, the generally accepted standard for production is eac recording + lame compression. Those who are experienced in such production process will figure out some tricks and use different parameter and parameter settings for different music. The compression ratio varies from the standard 128 kbps to the maximum of 320 kbps, but what is the difference and the difference in effect between these bit rates? ? How is the most suitable compression ratio, which one should be better for cbr and vbr etc. These topics are often discussed by everyone. Let me share with you some of my feelings.
The repertoire selected for this test is the first track of Bach’s “Grandenburg Concerto”, performed by the Munich Bach Orchestra, eac track recording software, cd’ex compression software, fooba2000 v0.8 playback software and listening earphones are er6 from Intech and e3c from Shure. Because the classical repertoire is very detailed and the band is large, the requirements for all aspects of sound quality are relatively high, so it can clearly reflect the difference in detail between different processing methods.
I first grabbed the track with rac, and then used the lame mp3 encoder (vision 1.92 engine 3.92) engine in the cd’ex software to process the wav file. I tried the lick parameters one by one to choose a good effect:
The first thread priority parameter selects the highest and lowest respectively. When other parameters are the same, the compression comparison shows that the degree of thread priority has no effect on the sound. The size of the generated files is the same, and the comparison sounds the same, so these parameters have no effect on the sound quality.
The second parameter is the version, which can be selected between mpegI, mpegII and mpegII.V. Similarly, the other parameters are determined and these three options are used to compress three times. After listening, although the file sizes of the three methods are all the same, but the actual listening feeling of mpegI is better. The mid-low frequency compression ratio is a bit smaller, but the high frequency distortion is a bit more. It is more suitable for listening to human voice and pop music. It is also good to use mpegI type to listen to classical music, the sound background is better, but if it is solo music with a lot of mid and high frequencies like violin, it is recommended to use mpegII.v type, which will have better results.
The third parameter is the most important, which is the bit rate. Choosing it directly affects the size and listening experience of your mp3 file. The higher the compression ratio, the higher the distortion, and the lower the compression ratio, the lower the distortion, but how do we find one for ourselves? What is the acceptable balance between the two? This requires careful exploration in the experiment. Considering that the sound quality of low bitrate files is not suitable for playing music, the minimum set is 128kbps, and four fixed bitrate files of 128, 192, 256 and 320 are used for comparison. and try.
The compression ratio of 128 kbps is still relatively rough, and the high-frequency part has obvious distortion after compression. It sounds hollow, wrinkled, scratchy, and often has a flickering sound. Misunderstanding, the compressed volume of a 3 minute 39 piece of music is 3414kb, although the volume is not large, the sound is not satisfactory, and there is a relatively large flaw.
192kbps bit rate compression effect is much better than 128. First of all, the sound is solid, at least there is no empty feeling, the high-frequency distortion is also much less, the sound is compact, the noise is small and clean, achieving relatively ideal listening. The sound effect, only because the compression is still relatively strong.

The relationship between MP3 frequency, bit rate, bit rate and sound quality

The relationship between MP3 frequency, bit rate, bit rate and sound quality.

mp3 encoding

Each song is ripped from a CD, converted to a WAV file, and then converted to MP3 using encoding software.

mp3 encoding lossy compression

So it should be a sample rate of 44100 KHz. Unless yours isn’t a song, but you record it as a WAV file and choose another sample rate when recording.
The main factor that affects the sound quality of MP3 is the bit rate. Now the best is 320K CBR (fixed bit rate) and VBR (variable bit rate), VBR files are a bit smaller than CBR. 192K VBR is the most popular on the Internet, which can meet the requirements of sound quality and file size at the same time, but I usually use CD to rip tracks or download APE (lossless compression, which can be restored to WAV file) and then convert it to 320K VBR.
Final reminder: MP3 transcoding is distorted and this distortion cannot be reversed. That is, if you convert MP3 to WAV sound quality, the file size increases dozen times, but the sound quality remains the same as MP3 sound quality.
If you want to hear low distortion, it’s better to listen to a CD or download APE.
First of all, sound quality is a very subjective thing!
It is often said that the sound quality is good, one refers to the good degree of reproduction, that is, the smaller the difference with the recording, the better; the other refers to the pleasant sound, which is good. As for mp3, mp3 is a compressed format, the higher the bitrate, the less compression and less loss of detail, that is, the higher the bitrate, the closer to the original sound. But sound quality is also related to your output device, such as a good mp3 player and a good pair of headphones, all of which will help your listening quality!
So if you want to improve sound quality, you can also start from the above perspectives and not overemphasize any one of them. When you have higher requirements for sound quality, you can give up mp3 and directly switch to stop CD. The CD carries waveform files, which are completely in lossless sound quality format, which will give better results.
If you want to reduce distortion, the only way is to increase the bitrate. It’s best to use variable bit rate (VBR) compression to produce mp3 files, which can strike a balance between maximum fidelity and minimum file size.
Finally, if you want completely lossless sound quality, you should still use audio files in a lossless compression format or an uncompressed file format. How good is the sound quality in MP3 format? 128/192/256/320 etc What is the difference in MP3 sound quality of various compression ratios/compression modes? What are some basic principles? How about the sound quality of other formats like APE/WMA/etc?

Bit rate – definition

Bitrate

Introduction

BITRATE

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 value for mobile phone stereo MP3 player, best setting value 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 dynamic bit rate (variable bit rate) means there is no fixed bit rate. 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.

Audio Coding Part5

VBR

About VBR

VBR Encoding

VBR: An interesting feature of MP3 files is that they can be read and played, which is also in line with the most basic features of streaming media. That is, the player can play without first reading the entire content of the file and play where it reads, even if the file is partially damaged. Although mp3 can have a file header, it is not very important for mp3 format files. Because of this feature, each frame of an MP3 file can have a separate average data rate without a special decoding scheme. That is why there is a technology called VBR (Variable bitrate, dynamic data rate), which allows each segment or even each frame of an MP3 file to have a separate bitrate, the advantage of this is that the sound quality is guaranteed to the maximum. . File size is limited. The advantages of this technology are obvious, but it is really difficult to use, because it requires the encoder to know how to assign the bitrate to each segment, which is like a dummy for encoders without waveform analysis. As such, VBR technology didn’t seem glamorous as soon as it appeared.
Experts have found that the human ear has a protective effect through long-term acoustic research. The sound signal is actually a type of energy wave, which propagates in air or other media. The most direct response of the human ear to the amount of sound energy, that is, the volume or pressure of the sound, is to hear the size of the sound. We call it the volume, which means the volume. The unit of energy is the decibel (dB). Even sounds of the same volume can be perceived by people as different in size due to their different frequencies. The 500 Hz frequency is most easily heard by the human ear. No matter whether the frequency is increased or decreased, even if the volume is the same, everyone will feel the sound become smaller. But when the volume drops to a certain level, the human ear cannot hear it, and each frequency has a different value.
You can see that this curve basically forms a V. When the frequency exceeds 15000 Hz, the human ear will feel that the sound is very small. Many people who are not very good at hearing cannot hear the frequency of 20000 Hz at all, no matter how loud it is… When the human ear hears two sounds with different frequencies and different volume at the same time, the one with the lower volume will also be ignored. For example, it is hard to hear the sound of the computer cooling fan during the day, but it becomes a noise source at night. According to this principle, the encoder can filter out many inaudible sounds to simplify information complexity and increase the compression ratio without significantly reducing sound quality. This shading is called the simultaneous shading effect. However, sound A is protected by sound B. If A is within the protection range centered on B, the protection will be more obvious. This range is called the critical bandwidth. The critical bandwidth of each frequency is different and the higher the frequency, the larger the critical bandwidth.
Frequency (Hz) Critical Bandwidth (Hz) Frequency (Hz) Critical Bandwidth (Hz)
Based on this effect, the experts designed a mental model of human hearing. After this model was imported into mp3 encoding, it led to a momentous revolution in sound quality. gradually eluted. At this point, the VBR technology, which has been buried for a long time, shines brightly, and with the use of the psychological model, it can perform powerful temptation and lethality.
For a long time, many people have a bad impression of MP3. More and more people think that the best sound quality of WMA is better than MP3. This statement is not correct. At medium and high bit rates, properly encoded MP3 is much better than WMA. It’s close to CD quality, with not-so-great hardware support, not many people can tell the difference between the two, it’s not a fairy tale, though you used to be able to easily tell the difference between MP3 and CD blindly. listening, but now cannot guarantee that it can distinguish correctly. Because MP3 is an excellent codec that was buried before.

What is the capacity of the high resolution sound source?

Hi-Res audio

High resolution sound source with more information than conventional CDs.

HiRes Audio

Since the data size is large, you can enjoy high-quality sound with a three-dimensional effect, but the problem is that when managing multiple high-resolution audio sources, the required storage capacity becomes huge.

Then I will introduce what is the capacity of the high resolution sound source, including the management method.

What is the capacity of Hi-Res Audio sources compared to CDs?
In determining the capacity (file size) of a high-resolution sound source, the sampling frequency and bit depth of the sound source are important factors.

The sample rate (sample rate) is a numerical value that is used as an index when converting analog data, such as speech, to a digital signal.

It indicates how many times per second an information sample was measured, and is expressed in “Hz (hertz)”.

If sampling is done every 44,100 seconds, it will be “44.1 kHz”.

On the other hand, the bit depth is a numerical value that indicates how many pieces are recorded in each divided data.

It is represented by “bit”.

Both the sample rate and the bit depth mean that the higher the number, the more information there will be, that is, the higher the resolution.

The amount of music data per second is called the bit rate.

Bit rate is the sample rate multiplied by the bit depth and is expressed in “bps”.

The calculation formula is as follows.

Bit rate (bps) = sample rate (Hz) x bit depth (bit) x 2

For example, the bit rate and sample rate of a CD sound source are generally “44.1 kHz / 16 bits”.

Most so-called “CD sound quality” sound sources are based on this number.

The size of the 5 minute 44.1 kHz / 16 bit / sound source file is about 50 MB.

But what about hi-res audio sources?

High resolution sound source capacity per song (5 minutes) varies depending on the music data format, as shown below.

WAV: 192 kHz / 24-bit: capacity for 5 minutes is 330 MB
FLAC: 192 kHz / 24 bit: capacity for 5 minutes is 200 MB
ALAC: 192 kHz / 24 bits: capacity for 5 minutes is 200 MB
What you can see from this is that the capacity of the high resolution sound source is 4 to 6 times that of the CD sound source in 5 minutes.

Large capacity high resolution music management equipment
If you download 5 high-resolution songs for 4 minutes, it will take up about 700MB (for 96kHz / 24-bit WAV files).

In the case of 10 songs, it exceeds 1400MB, that is, 1GB.

If so, I would like to have enough storage to handle that amount of data.

An effective way to do this is to build a NAS-centric network audio system that incorporates a large-capacity hard drive.

For example, if you can prepare a 4TB (terabyte) hard drive, it can store around 20,000 high-resolution songs.

Next, we will explain what NAS and network audio are like.

NAS
NAS stands for “Network Attached Storage” and it reads like aubergine.

It stands for network attached storage, and it is also called a network hard drive or network compatible HDD.

In other words, it is an external hard drive that is used when connecting to a network (LAN).

A normal external hard drive used when connecting to a PC = PC via USB etc. can basically be used with only one PC.

However, if it is a NAS, it can be used with multiple devices participating in the LAN.

Files saved on the hard drive can also be used and shared from, for example, the personal computers of each family member participating in the home LAN, smartphones connected via Wi-Fi, and TVs in the living room to be.

It is also possible to access the data on the NAS from the outside via the Internet.

The NAS is often used in the home to store and share data for music, video (video / TV recording data), photos (images), etc.

Meaning and relationship of sample rate, bit depth, and bit rate

bit depth

Sampling rate
Bit depth
Bit rate

bit depth audio

I will present the three meanings and relationships of.

Table of Contents
What is the sampling rate?
What is bit depth?
What is a bit rate?
resume
Sponsorship

What is the sampling rate?
For example, let’s say you say “Ah” for 1 second.

When recording this “Ah” sound on a personal computer, the “Ah” sound is divided into tens of thousands per second, each divided into tens of thousands.

“The height of this section was about this.”
“The length of this section was about this.”
Record it as data like this.

The personal computer continually reads each of this divided data and outputs it as a “voice” called “Ah”.

At this time, “how many tens of thousands of sounds are collected per second” is called the “sample rate.” (Also called “sample rate”)

Sampling rate
▲ Sample rate image

The more divisions you make, the smoother the sound will be, and as a result, you will feel that the sound quality has improved!

What is bit depth?
The sample rate is “how many tens of thousands of sounds are collected per second”.

“How much capacity is given to each divided data (sample)” is called “bit depth”.

Figure_bit depth
▲ Bit depth image

Also called “number of quantization bits”, “number of sample bits”, “bit offset”, and so on.

For example, if the bit depth is “16 bits”, the amount of information is 2 to the 16th power (65536) for one sample.

The higher the bit depth, the greater the expressiveness of the sound’s fineness and volume, and as a result, I feel like the sound quality has improved!

By the way, the bit depth of most of the world’s sound sources is 16 bit.

that’s why

“Import music from CD!”
“Import music downloaded from the Internet!”
In such cases, 16 bits is sufficient.

On the other hand, if you say “What you recorded in your DAW comes out wav!”, It is better to have 16 bits or more.

This is because, for example, when processing audio effects with audio editing software, sound deterioration can be reduced to zero by assigning an additional bit depth (for example, 32-bit). (Although 16 bit is fine for final output)

What’s more,
note that “bit depth” on this page has a different meaning than “bit depth” in video.

Reference: What is Bit Depth (Color Depth)? Differences like 10bit / 24bit / 30bit

What is a bit rate?
Bit rate is the “amount of data per second”.

Reference: What is a bit rate? Relationship between image quality, sound quality and codec [Video / Audio]

The “sample rate” and “bit depth” presented above are

Sample rate: how many tens of thousands of sounds are collected per second
Bit depth: how much to give to each divided data
Therefore, the product of these two values is the “bit rate”.

Audio encoding

Audio Encoding

I wrote over audio files last time, but if you reduce the file size (code at a lower bit rate), the sound quality tends to deteriorate. How much should it really be? .. ..

audio encoding

When compressing using audio encoding (AAC, MP3, etc.), the compression rate is determined by the bit rate at the time of encoding. Specifically, if you set a low bit rate, the compression rate will be higher and the file size when saved will be smaller, but first of all, what is the bit rate for uncompressed original sound source (PCM) ?
If you save it as PCM, the sound quality will be that of the original sound, but it can be a bit awkward to save without worrying about the file size. Also, depending on the application, I think the memory capacity is sufficient even for the original sound size and the communication speed is fine. Therefore, I would like to write about the sample rate and bit rate that are often heard in digital audio.

The bit rate of digital audio is determined by the sampling frequency, the number of bits assigned to a sample (number of quantization bits), and the number of channels (stereo, monaural, etc.).

PCM bit rate (uncompressed) = sample rate x number of quantization bits x number of channels
As I wrote a bit last time, in file containers like wav and mp4 format, this information is attached as a header, so that the application can see the header and play it back. The compression rate of the encoding is determined by the bit rate specified at the time of encoding for this PCM (uncompressed) bit rate.
For example, as many of you know about music CDs, with 44.1 kHz stereo, this is the next bit rate.

Music CD bit rate: 44100Hz x 16bit x 2ch (stereo) = 1411.2kbps
When encoding this with MP3, AAC, etc., it is natural to specify a bit rate lower than 1,411.2 kbps. For example, when encoding at 256 kbps, the compression rate is around 18% when the original sound is 100% and the file size is 1/5 or less.

Encode a music CD at 256 kbps: 256 kbps / 1,411.2 kbps = about 18%
In general, the sample rates of audio devices connected to PCs are 48 kHz and 44.1 kHz for music, 16 kHz and 8 kHz for audio such as microphones and headphones, and 32 kHz, 24 kHz, 22.05 kHz. , etc.

The bit rate of PCM (uncompressed sound source) with 16-bit quantization bits is as follows.

Stereo (for music) PCM 16-bit bit rate (example)
Sampling frequency Number of quantization bits Number of channels Bit rate Comments
48kHz 16 16 2 1536 kbps
44.1 kHz 16 16 2 1,411.2 kbps Music CD
32kHz 16 16 2 1,024 kbps
24kHz 16 16 2 768 kbps
22.05 kHz 16 16 2 705.6 kbps
Monaural (for audio) PCM 16-bit bit rate (example)
Sampling frequency Number of quantization bits Number of channels Bit rate Comments
32kHz 16 16 1 512 kbps Super Wide Band
24kHz 16 16 1 384 kbps
16kHz 16 16 1 256 kbps broadband
8kHz 16 16 1 128 kbps Narrow band

Sampling rate
If you check the web, there are explanations such as the sampling required to convert analog waveforms to digital conversion. For example, it shows how many samples of an audio signal input from a microphone are taken per second and digitized. The larger the sample, the greater the range that can be recorded. When an analog waveform is digitized, the frequency that can be expressed is half the sampling frequency (sampling theorem). For example, with a sample rate of 48kHz, it is possible to express up to 24kHz. At 8kHz (narrowband) and 16kHz (wideband), which are often used for audio, you can only hear up to 4kHz and 8kHz, respectively. The higher the sample rate, the higher the bit rate.

sampling theorem
It is a very simple explanation, but it can express up to half the sample rate. When sampling a signal, if the interval is small, it can be restored close to the original signal, but if it is too thick, it cannot be restored (I would like to write a little more detail when talking about signal processing or other time ).

What is the bit rate? Simple explanation

bitrate

Bit rate is a unit of data transfer.

When used in video or audio, as in video editing, it indicates how much data is represented per unit of time, and “bits per second (bps)” is generally used.
A bit is the smallest unit of data that a computer handles.
Two states of “0” and “1” can be expressed by 1 bit. (1 binary digit)
Similarly, a byte representing the size of a file is a unit of data handled by a computer. (1 byte = 8 bits)
Here are some things to keep in mind about bit rates.
In home appliance hard disk recorders, it is sometimes expressed as a recording mode such as XP, SP, LP.
The higher the bit rate value (numerical value), the better the picture and sound quality, but the greater the amount of data (file size).
The unit of bit rate is usually Mbps, which means 10 6 (10 to the sixth power) for video, and kbps, which means 10 3 (10 to the third power) for audio.
When burning a video to DVD, there is a limit to the amount of data that can be burned to DVD, and if you try to burn a long video, you will have to lower the bit rate, resulting in poor image quality. . To record high-quality video, the bit rate must be increased, which increases the amount of data required and shortens the recording time.
Even if the bit rate used is the same, the image quality and sound quality will differ depending on the encoder and compression method used for digitizing.

What are “bit depth” and “sample rate”? Part 2

Understanding Sample Rate, Bit Depth, and Bit Rate - Headphonesty

What is the sample rate?

Bit Depth

Next, I will explain the sample rate.

The sample rate is like the “resolution” of the audio.

The higher the sample rate, the more samples per second = you can hear better.

Requires double sample rate

One thing to keep in mind here is that you need twice the sample rate to hear sound at that frequency.

For example, if you want to hear a 1000 Hz (1 kHz) sound accurately and clearly, the sampling frequency must be at least 2000 Hz (2 kHz).

If the sample rate is less than twice the value you want to hear, “aliases” will occur and you will not be able to process the sound accurately, such as crackle or noise.

Nyquist frequency

By the way, to use a little technical word, it also means not to exceed the “Nyquist Frequency”.

The Nyquist rate is exactly half the supported sample rate.

For example, if the sampling frequency is 44.1 kHz, the Nyquist frequency will be 22.05 kHz.

If you try to handle this high-pitched sound that exceeds 22.05 kHz, the above-mentioned “aliasing” will occur and you will not be able to reproduce the sound correctly.

Range recognizable by the human ear

The loudest sound that can be recognized by the human ear is said to be 20 kHz, so to hear a 20 kHz sound, you only need to have a sample rate of at least 40 kHz.

After that, to avoid aliasing, apply an anti-aliasing filter until the Nyquist (Transition Band) frequency is reached.

For 44.1 kHz, 2050 samples x 2 are required.

In other words, a sample rate of 44.1 kHz is all that is needed to minimize the limit of sound (20 kHz) that the human ear can hear.

When the sampling frequency is high (96 kHz, 192 kHz)

Recently, it can be set to a high sample rate, such as 96 kHz or 192 kHz.

Unfortunately, even with such a high sample rate, it’s hard to tell the difference.

As shown in the image above, non-human animals can hear higher frequency sounds.

However, it is a level that we do not have to worry about because it is a completely inaudible zone for the human ear.

By the way, many audio interfaces cover up to 192 kHz.

Controversy over sampling rate

In fact, in the 1970s, many media outlets were controversial about sample rates.

At the time, 48 kHz was the audio standard used in radio, television, and video work.

However, broadcasting stations have decided to use 44.1 kHz as a standard to prevent data from being copied to consumers (viewers) by intentionally breaking compatibility (or making conversion difficult) …

It’s difficult to change data from 44.8 kHz to 44.1 kHz, so it prevented the average viewer from converting it to the sample rate used for home devices.

By the way, the article “Comparison of professional versus cheaper audio interface” is summarized here, which is also explained from the point of view of bit rate and sample rate.