Tag: what mp3 quality is best

Free Download Mp4Gain

Why does the sound change even though it is digital?

If you’re interested in PC audio, once you think about it, “Why does the sound change even though it’s digital?

It is unlikely that the data cannot be corrected and returned to the original data with current digital audio equipment, so the digital data itself is safe to think that they are the same. Then why?
The reason is that even though it is digital, the signal that drives the speaker is an “analog signal”, so it affects the “analog signal” after the digital is converted to analog.
Suppose you have a noisy PC. As shown in the figure below, noise flows through the GND of the USB cable to the USB-DAC. The noise flowing into the USB-DAC affects the signal after analog conversion and the sound quality deteriorates.

On the other hand, a PC with less noise has less effect, so the sound quality is less likely to deteriorate.

In the case of “wireless” at the end, “wireless” is the “ultimate USB cable”. It is not affected at all by PC noise. It is an ideal transmission method. The only embodiment of such a method is “Air High Resolution Link Technology”.

Free Download Mp4Gain


Mp4Gain Main Window


Mp4Gain Features


Free Download Mp4Gain

What is PC audio?

For super beginners

Does anyone say “I really don’t understand” or “I can’t hear you anymore” over the audio? This is a super introductory course on audio that started in response to so many unexpected voices, and this time I will explain very briefly about the “PC audio” that I often hear recently.

SHARE
● First of all, what is “PC audio”?

This time, let’s introduce PC audio.

Until a while ago, the only way to listen to music was with a CD, MD or radio player.

However, before I knew it, listening styles with digital music players, such as personal computers and iPods, became common.

And the idea of ​​PC audio is “I want to listen to music with good sound when I listen to music using a personal computer.”

● The point is “DAC” which performs the digital to analog conversion.

For example, when you listen to music on your computer, you usually listen to it like this.

Headphones for PC.png

Connect headphones, etc. to the audio output terminal (LINE OUT) of your computer.

Or you can play it through your computer speakers.

So how do you make this sound good?

When playing music with digital data, the DAC (duck) circuit is important in terms of sound quality.

Since MP3 or WAV is digital data, but scanning for speakers or headphones to play, you need to convert the digital data to analog signal.

DAC (short for D / A converter) performs the “digital-to-analog conversion”.

The quality of the converted analog signal depends on the accuracy of the DAC.

In other words, if the DAC is not good, it will not sound good at all.

Originally the digital data is a list of numbers like “0101001 …”, but the figure is as follows. (* Image)

DA .png conversion

If you are using your computer’s audio output terminal, you are using the DAC circuit inside your computer as shown in the figure below.

However, since the personal computer was not originally made for music playback,

The performance of the DAC installed in the personal computer is often not that high due to the balance of costs with other parts.

Also, many electronic parts run at high speed inside the personal computer, so a lot of noise is generated.

In such an environment, you cannot expect “good sound” in the audio sense.

PC Headset (with DAC) .png
Therefore, instead of using the DAC inside the personal computer, I asked them to send the digital data as it is from the USB terminal.

It is converted to analog with a music DAC and converted to an audio quality music signal.

That’s the basic idea of ​​PC audio.

Is it natural that “the sound of the computer is bad”?

Compare the performance of the audio output numerically

If you usually use an audio interface or USB DAC, many people think that the sound function installed on the PC is obsolete. The author himself thought so, but “actually, the sound quality may have improved with recent PCs.” Suddenly, I thought about that, and this time I checked the sound quality of the computer I had.

As long as you listen to a little sound with headphones, you won’t get the bouncing buzz like in ancient times. Although it is clearly different from listening to solid equipment, I don’t think there is any particular discomfort in listening to streaming audio over the Internet with a PC alone without listening to it. So, I used RMAA PRO, which I always use to check the audio interface, to compare the difference numerically. Then there was a considerable difference between the models I tested, so I summarized the results.

Compare barebones with notes and Intel NUC
I believe that an audio interface and USB DAC are a must to produce decent sound on a PC, but even professional musicians often connect to the PA from the PC’s headphone output without using the audio interface. Sometimes I see people. Well, I was wondering if I shouldn’t be particular about the sound, but I was wondering, “Is it okay?”

However, there is a chance that the onboard sound function is now decent. It is not a good idea to decide that it is not good without verifying it. So I decided to test it with three relatively new PCs that I had.

One is a basic “Shuttle SH370R6” PC that was purchased late last year and assembled early in the new year. I wanted to incorporate Intel’s Core i9, but since it only supported the 8th Gen Core processor, it was a machine with 6 cores, 12-thread Core i7-8700 installed, and 32GB of memory.

Barebone PC “Shuttle SH370R6”
After that, it seems that the 9th Gen Core processor also supports BIOS update, and I am sorry I bought it a bit later, but I will test it on such a machine.

Uses Core i7-8700 with 6 cores and 12 threads
The second is a small notebook PC that I bought last fall and the NEC “LAVIE Note Mobile NW150” that I usually carry in my bag every day. A lightweight machine weighing 904g with a battery that runs for 13 hours on an 11.6-inch screen. The CPU is a 2-core Pentium Dual-Core 4410Y that runs at 1.5 GHz, so it is not a fast machine, but it is a portable PC that is enough to surf the net with Chrome and write manuscripts with an editor. of text.

NEC 「LAVIE Note Mobile NW150」
The third is an Intel NUC kit that I bought two years ago and it is a small machine called “7i7BNH”. It is a PC that I bought because it is equipped with a 7th generation Core i7-7567U and a USB Type-C type Thunderbolt 3 terminal.

Intel NUC 「7i7BNH」
Typically, if it is an audio interface, the input and output are connected directly with a cable to make an audio loop, and then the measurement is done using the RMAA PRO audio testing software. I’ve always tried 44.1 kHz, 48 kHz, 96 kHz, and 192 kHz, so I thought I’d use that method again … but if you look closely, it has input and output terminals. Only the first SH370R6. Both the LAVIE Note Mobile and the NUC have only one headphone-out jack.

Looking around now, notebook PCs are less likely to have LINE IN and mic input, and more common to have only output. Also, the purpose of the experiment here is not to check the total input / output function, but to see what the output performance of the headphones is like. In a normal audio loop, if there is a problem with the input = record function, that becomes a bottleneck, and even if the headphone output is high-performance, it will be poor. So I decided to use the same audio interface for all inputs and compare the output performance of each headphone.

This time, I used Roland’s Rubix 24 as an audio interface. It is a 2IN / 4OUT audio interface that works with USB bus power and is a device that can record and play back at a maximum of 192 kHz / 24 bits. First of all, the result of testing with RMAA PRO in the form of an audio loop to see the performance of this Rubix 24 itself is as follows. I have tested the 2IN / 2OUT Rubix 22 before, and the results show that it has roughly the same performance.

Incredible sound quality digital reverb!

Amazing sound quality digital reverb thanks to high-speed computation using high-precision 32-bit DSP!

AMBI SPACE reproduces the complex reverb mechanism created by various elements with the original FREE THE TONE algorithm, making full use of high-speed arithmetic processing using a 32-bit dual-core CPU chip and high-precision 32-bit DSP. bits. The high-quality, yet extremely natural, musical reverb produced by this small enclosure can be used in a variety of situations, not only for electric guitars and basses, but also for acoustic guitars and vocals, and even for recording mixes. Additionally, FREE THE TONE’s original new reverb sounds “CAVE” and “SERENE” emphasize the frequency components by reproducing the early reflection and late reverb that make up the reverb. By adding the complex harmonic structure produced by the above to the multi-stage reverb sound, we create a reverb sound that is fantastic, transparent and expansive like never before. It is a reverb effector that was born after many years of development combining the analog and digital technologies that Free The Tone has cultivated.

“Characteristics”
・ By using a dual-core chip with a 32-bit main CPU and a 32-bit coprocessor and performing high-speed calculations with a high-precision 32-bit DSP, sound comparable to that of a rack-type effector is achieved by being the size of a compact effector.
-Up to a total of 4 presets for all knobs can be stored as presets.
-Four presets can be recalled using MIDI program change numbers.
-Equipped with 6 reverb MODES.
-Free The Tone’s original reverb sounds “CAVE” and “SERENE” are illusions that have never been seen before by adding sounds with various overtone structures created by a complex reverb pattern design to the multi-stage reverb sound. Produces a reverb sound that is both transparent and spacious.
-The original sound (dry sound) passes from the input to the output as an analog signal, and the original sound and the reverb sound are mixed by the internal analog mixer. This will come out without compromising the sound quality of the original sound.
-Built-in HTS circuit that comprehensively manages input to output signals, which is an important feature of FREE THE TONE products, keeps the sound texture when the effect is on and off the same.

《Main Specifications》
● Number of presets: 4
● Input impedance: INST 1 MΩ or more / LINE 300
kΩ or more ● Output load impedance: 10 kΩ or more
● Maximum input level: INST + 3 dBm / LINE + 11 dBm
● Control: PRE DELAY, DECAY, TONE, MIX, MODE, INST (-10dB) / LINE (+ 4dB) level change switch, KILL DRY switch
● Terminal: standard 1/4-inch x 4 L (mono) / R INPUT phone jack, L (mono) / R OUT, DIN 5PIN (MIDI IN) jack, DC9V input jack (for connecting AC adapter )
● Power supply: DC9V dedicated AC adapter (FA-0905D-JA)
● Current consumption: 280mA (maximum value)
● Size: 120 (W) x 102.3 (D) x 74 (H) mm (including projections such as the foot switch and connector)
● Weight: Approximately 385 g (without
accessories) ● Accessories: Warranty card, instruction manual, dedicated AC adapter (FA-0905D-JA), rubber feet x 4

High Resolution Audio Source

It can be said that “USB-DAC” is a secret weapon for playing music files on a computer with high sound quality.

Just add “USB-DAC” to the audio you use all the time, and you can enjoy much higher sound quality! Therefore, this time, Sara-chan visited Onkyo Co., Ltd., which developed the state-of-the-art “USB-DAC” that supports high-quality sound sources called “high-resolution”, which has been launched more and more in the last years. ! We also visited the audition room and asked Mr. Kurosawa, director of the high-quality music distribution site “e-onkyo music”, to teach us how to enjoy high-quality sound!
“Sample rate” and “bit rate”

Sara-chan: Hello! Wow, it’s a nice listening room! I’m excited!

Kurosawa: Hi Sarah! Today, I am trying to get you to experience high-end sound quality in various ways.

Sara-chan: Thank you! “USB-DAC” is certainly important to enjoy high quality music on your PC! But there are many “USB-DACs” and I don’t know what to choose.

Mr. Kurosawa: When choosing “USB-DAC”, it is a good idea to check the “sample rate” and the “bit rate”.

Sara-chan: Sa, Samp … Call frequency ?? What the heck is that ~ ??

Mr. Kurosawa: “USB-DAC” is a device that converts sound from digital signals to analog signals, isn’t it? The “sample rate” indicates the number of digital samples of the audio signal acquired per second during the conversion. The “sample rate” determines the frequency range of the audio file. The higher this number, the closer the digital waveform will be to the original analog waveform and the softer the sound will be. On the other hand, “bit rate” indicates the amount of information per second. They are expressed in units of “Hz” and “bit” respectively. By the way, do you know what the CD standard is?

Sara-chan: Well I’m sure it’s “44.1 kHz / 16 bit”!

Mr. Kurosawa: That’s right! It is said that the sound in the ultra high range above 20 kHz cannot be heard by the human ear, so the CD cuts out the inaudible sound. But even if you think you can’t hear it, you actually feel the vibrations in the air and it affects the sound at the frequencies you hear.
Since the high resolution sound source also records that part, it can be said that it is closer to a more realistic sound. First, at the music creation stage, work is often done at 96 kHz / 24-bit, which is why high-resolution sound sources are sometimes referred to as “studio master quality.” Recently, even more informational sound sources such as “192kHz / 24bit” have appeared. There are “44.1 kHz / 16 bit” and “96 kHz / 24 bit” sound sources here, so let’s compare them using ONKYO’s “DAC-1000 (S)”!

Sara-chan: Wow! You can feel the difference more than you imagined! It feels like a live performance is taking place right in front of you! I feel that the sound is expansive and I feel that I am surrounded by the sound! Anyway, it seems like I’ve never heard it on audio before! Impressed!

Kurosawa: Fufufu. You can feel the difference! However, even if you have a high resolution sound source such as “96 kHz / 24 bit”, there is no point in using a “USB-DAC” that does not support “96 kHz / 24 bit”. Therefore, when choosing “USB-DAC”, it is important to check the “sample rate” and “bit rate” to see if it is compatible with the sound quality you want to hear. By the way, recently there is even a “USB-DAC” that has a function to change the frequency called “upsampling”.

Sara-chan: Wow! We are in an era where high quality sound is increasingly required!

What is asynchronous forwarding?

Sara-chan: What other characteristics should I check?

Mr. Kurosawa: The data transfer method is important! When transferring a high quality sound source from a personal computer to a “USB-DAC”, some have a function called “asynchronous transfer” to avoid data corruption.

What is sound source compression?

In the old days, you used to put a CD-sized Walkman and a CD case with your favorite CDs in a bag and go out.

Today, you can carry a large number of songs in the palm of your hand and enjoy music anytime, anywhere. The evolution of the times is amazing. Why is it possible to do this? It’s all thanks to “audio compression”!

Why compress in the first place? Doesn’t the sound quality get worse?
Music data can generally be burned to CD (700MB) for only 80 minutes without compression, but when compressed, the data size is reduced to about 1/10, which is 10 times larger in a place with the same capacity. capable of recording music. In other words, up to 10 music CDs can be burned onto one CD. Also, you can attach emails and download them from the Internet thanks to “audio compression”!

The sound quality is annoying, but it is not 100% the same, but it can maintain the same sound quality as an audio CD.

“Audio compression” has become an indispensable part of our musical life. There are various audio compression methods like MP3 and AAC, but I think there are many people who say, “I really don’t understand the difference …”.

First of all, from the basics of music files.
Music files are roughly classified into three types: “uncompressed”, “lossy compression”, and “lossless compression”. The nature of each format is as follows.

■ Uncompressed The
Sound quality is good because it is not compressed, but the file size is large.
<File format: WAVE (WAV) / AIFF>

■ Lossy compression When
compresses with lossy compression, it cannot be undone and the sound quality deteriorates, but the file size can be reduced.
<File format: MP3 / WMA / Ogg Vorbis / AAC / AC3 ​​/ ATRAC3 / ATRAC3plus / RealAudio / TwinVQ>

■ Lossless compression
Even if it is compressed once, it can be restored to the original data, so there is no deterioration in sound quality. The file size will be a bit larger, but recently the capacity of the PC hard drive has also increased, which is why it is popular.
<File format: Flac / Monkey’s Audio / TTA / WMA Lossless / RealAudio Lossless / Apple Lossless>

What are the MP3s that you see often on the Internet?
It uses a lossy compression method that achieves a high compression rate and is an audio compression method that compresses the amount of data to approximately 1/11 (128 kbps) while maintaining the sound quality of a music CD. In other words, the voice is compressed cutting the data in the part that is difficult for humans to perceive. The original data is compressed with the idea that “this part cannot be heard accurately by the human ear, so cut it out!”
What is the AAC you hear often these days?
AAC is a relatively new audio compression method. ITunes uses it in familiar places. AAC has a compression efficiency 1.4 times higher than MP3 and the sound quality is almost the same. The name itself may not be heard as much as MP3, but it is actually used in many places. For example, QuickTime, BS digital / 110 degree, CS digital / digital terrestrial / single segment broadcast, “Chaku-Uta” mobile phone, and so on.

Various audio compression methods
In addition to MP3 and AAC, there are several methods of audio compression.

■ Uncompressed
WAVE
(WAV)
.wav WAV or WAVE (RIFF waveform audio format). A format for writing audio data developed by Microsoft and IBM. A file format used primarily by Windows.
AIFF .aif A file format for audio data developed by Apple Computer, the standard audio format for Mac. Mac version of WAVE.

■ Lossy compression
MP3 .mp3 The compression format that has been overwhelmingly popular until now. It uses a lossy compression method that achieves a high compression rate by reducing data that is difficult for humans to perceive. Downloading music from the Internet, etc.
WMA .wma Abbreviation for Windows Media Audio. It has a copyright protection function and is used for broadcast distribution. Like other major compression methods, it uses a lossy compression method that achieves a high compression rate by reducing data that is difficult for humans to perceive.
Ogg Vorbis .ogg Popular as a next generation format for MP3. It is a lossy audio compression standard similar to MP3, WMA, AAC, etc., but differs from existing standards in that it is a free and open standard. One of the main features is that there are fewer rights restrictions than other codecs and formats.

Digital sound and analog sound

The “sound” can generally be expressed as a “waveform”.

Analog recording is used to store this “waveform” as it is. Since the original waveform is stored as is, the sound quality appears to be good, but the analog data deteriorates in the playback process and the noise is mixed. However, since digital data does not deteriorate much in the playback process, it can be reproduced closer to the original sound without noise.

Also, if you copy an analog recording, the original waveform will gradually collapse and the sound quality will deteriorate. On the other hand, digital recording is recorded by dividing this waveform into small pieces and quantizing them. This value does not change even if you copy it, so the sound quality will not deteriorate. It can also be easily imported to a personal computer for processing.
Digital sound quality can be expressed in “bits” and “kHz”. “Bit” is the pitch of the sound volume. Also, “kHz” is the number of times a second is divided into digital data, and the higher the value, the higher the sound quality.
The must-have PCM recorder for musicians

“Linear PCM Recorder”.
It’s no exaggeration to say that no jazz musician doesn’t. When I go to a jam session or a live concert, I see all the members take out their PCM recorder and set it up.
I think this was MD a while ago, but why did musicians switch to PCM recorders? The first reason is that it is possible to record “good sound”. Let’s take a look at the history of music recorders and high-quality sound.
The Sony “Walkman” is the first most portable recording device.
It was initially used as a play-only cassette player, but after a few years it was equipped with a recording function.
After that, the era of MD arrived, and soon MD recorders were developed, and the clear digital sound quality, albeit compressed, gave a new step to recording. And DAT (Digital Audio Tape) is also a major player in the history of recording. DAT is a tape medium that enables digital recording. Immediately, a recorder using DAT was launched and its high-quality sound was well received. However, due to a copyright issue, DAT disappears.
After that, in the recording market, IC recorders focused on language study, lesson recording and conference proceedings appeared, but music recorders that sought high sound quality were never made.

However, with the recent band boom and the increasing music population, the need for music recorders has increased. In response to that, the PCM recorder appeared and attracted a lot of attention.
The biggest difference between a PCM recorder and a conventional recorder. It is in compression. Compression is the process of reducing capacity while preserving data content. In the past, most audio media was compressed and recorded. Sound quality deteriorated significantly due to compression, but there was a reason I kept using it.
This is because many semiconductor memories required for long-term recording were still expensive at the time.
However, in recent years, the price of semiconductor memory has fallen to the point that it can be kept at a reasonable price even if memory capacity is increased, leading to the birth of uncompressed linear PCM. By installing a high-capacity semiconductor, it was possible to record with good sound without compression.
24 bit / 96 kHz beyond CD
Some linear PCMs have a sound quality that exceeds that of a CD. That is the “24bit / 96kHz” format.
Of course, general CDs are not compressed. The format is “16 bit / 44 kHz”. What this means is that 16 bits for 44.1 kHz sampling, that is, 1 second is distinguished by 44,000 units, and from small to loud sounds in a unit of time it is a 16-bit gradation, that is, 56,000. It means that it is expressed in gradation. That is a CD.
On the other hand, there are several types of formats for linear PCM, but basically, the higher the sampling frequency and the higher the number of bits, the higher the sound quality and the better the sound can be recorded. It means that 96,000 times they can analyze sound more precisely than 44,000 times, and the analysis allows for more delicate expression of sound in 24-bit than 16-bit. Today, recorders with such amazing functionality have been marketed.

Techniques for enjoying compressed audio with high sound quality

I converted a music CD to MP3 or WMA, but the sound is not good … Many users will think so. So, let me introduce you to one of the methods to do high quality audio compression.

Sound quality is improved by lowering the volume of the extracted WAV file and then encoding it in MP3 or WMA.

There are various audio compression methods like WMA, AAC, Ogg Vorbis for MP3. They are very convenient because they can compress the file size to about 1/11 (at 128 kbps) compared to WAV. However, these are called lossy compression and sound quality degrades slightly because compression is done by reducing sound information to the point that it is difficult for the human ear to understand. If you really stick to the original sound of a music CD, you should use WAV as is, or use lossless (lossless) compression like WMA Lossless, Apple Lossless, FLAC.

However, even with MP3 and WMA, if it is compressed well, many people will not notice the difference in sound quality. Generally, a bit rate of 128 kbps is used, but if you increase it to 160 kbps or 192 kbps, the amount of data to be reduced will decrease and the sound quality will improve.

If you can’t tell the difference from the original sound by listening to the lossy compressed sound, that’s fine, but it is troublesome that there are cases where you can clearly see the deterioration depending on the song. Specifically, there is a clip-like beep, and the cymbal and hi-hat sound is clearly distorted. This is not just a problem to be solved by increasing the bit rate, it has nothing to do with the lack of high frequencies. This happens because the music data matches the weak points of the compression algorithm.

The weak point is that it is vulnerable to the sound applied by the compressor (equalizing the maximum volume). In other words, it is vulnerable to loud and continuous sounds. In particular, the recent J-Pop is made by applying a compressor to gain sound pressure and setting it to maximum volume, so it can be said that it is the music data that is vulnerable to compressed audio.

Therefore, if you process the WAV data a little before compressing it to MP3 or WMA, the sound quality will be drastically improved and clipping phenomenon can be avoided.

The process is simply to lower the volume level of the entire song. Of course, it’s burning, so you can’t mess with the volume of CD playback or squeeze the volume from encoding software like Windows Media Player. It is necessary to rewrite the waveform of the WAV data using waveform editing software and lower the volume level. Specifically, it is safe to lower the level by about -6dB. This will cut the volume in half, but due to the features of the compression software, it doesn’t appear to be as quiet. If possible, it is even more effective to apply an effect called an expander to eliminate the compressor effect.

Improve quality of mp3s Part 3

Be aware of the difference between bit rate and bit depth!
Bit rate: amount of information per second
Bit depth: amount of information per sample divided by the sample rate
In other words, the calculation is “bit rate = bit depth x sample rate”.
What values ​​can be set when exporting MP3? What is the best export configuration?

~ What items can be set when exporting MP3? ~

When exporting MP3, you can set the following two items 💡

Bit rate: 16 kbit ~ 320 kbit
Sampling frequency: 32,000 Hz, 44,100 Hz, 48,000 Hz
For example, if there are no specifications in a competition and you want to export with good sound quality with mp3, let’s export with “320kbit, 48,000Hz” 💡

What is the best setting to reduce capacity and export with good sound quality?
The capacity is small!
Sound quality is good!
So if you want the capacity to be as small as possible but also the sound quality as best as possible, which setting is better to export?

The sample rate is generally 44,100 Hz.
Use ~ 44,100Hz! ~

In the video industry, 48,000Hz is mainstream, but in music, the sound quality is high enough if it is 44,100Hz, which is used for CD.

In the blind test, it is said that about half of the people can distinguish between them and in addition, there are two options, so even if you answer properly, there is a chance that you will win about half.

The bitrate is around 128 kbit, which is the limit between good and bad.
~ 128 kbit or more is recommended! ~

Even if the bit rate is reduced to around 128 kbit, the print does not change much and the roughness does not appear.

If you lower it to reduce capacity, “about 128 kbit is a guideline” 💡

If you want to reduce the capacity and stick to the sound quality, it is better to select 128 kbit ~ 192 kbit and 44,100 hz to export.
~ 44,100Hz 128kbit ~ is the best!

For those who want to reduce capacity and focus on sound quality, it is better to set the sample rate to 44,100hz and the bit rate to 128-192kbit 💡

MP3s also have export settings. What are the settings for exporting with even slightly better sound quality? summary of
MP3 stands for “MPEG 1 Layer 3” compression method
Depending on the compression settings, the capacity may be reduced by 1/10 or more.
MP3 removes sound components that are inaudible to humans, thus maintaining sound quality.
Set “bit rate” and “sample rate” when exporting
The sample rate is “number of samples per second”
Bit rate is “amount of information per second”
Bit depth is “amount of information per sample”
The best sound quality settings for MP3 export are 48,000hz, 320kbit
44,100Hz, 128 ~ 196kbit is the setting that balances capacity and sound quality when exporting MP3.
MP3s are often used to check demos and save space 💡

Even if you have exported it casually, put it in the corner of your head that there is a setting for MP3 export, and when you need it, remember it and use it

Improve quality of mp3s Part 2

In other words, MP3 can be said to be a sound source file that was originally created by removing only the sounds that are difficult for humans to hear and without modifying the other sounds.

It is a commercially useful file format because it sounds very nice even if the capacity is small 💡

About MP3 Export Settings

When exporting MP3, the sound quality changes greatly depending on the bit rate (amount of data per second) and the sample rate (number of samples per second).

The sample rate determines how many divisions per second
~ How many divisions per second? ~

The sample rate determines how many divisions of information are handled per second.

At 44,100Hz, which is common, one second of information is divided into 44,100 samples, and at 44,800Hz, it is divided into 48,000 samples 💡

If you divide the sample into smaller samples, the information will be seamlessly connected, and if the information is approximate, it will be staggered.

Bit rate (kbit) determines the amount of information per second
~ Amount of information per second ~

On the other hand, the bit rate (bit) determines “the amount of information a sample has in one second”.

soon,

Bit rate = bit depth (1 sample information) x sample rate (number of samples per second)

For example, the bit rate of a 16-bit 44,100 hz wav file is 705.6 kbit.

Note that the bit rate is sometimes called “bps (bits per second)” because it is the number of bits per second 💡

caution!

* Bit depth is used in WAV export settings, etc. which is very similar, but the bit depth is the amount of information per sample. Be careful because it is confusing!