What is the sample rate (sample rate)?


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What is the sample rate (sample rate)?

Sample Rate

Differences in sound quality and how to check it! It also explains the settings to consider!

sample rate

Adjusting the sample rate Sound quality

The sample rate is mentioned in the DAW and audio interface settings. How should I set it?
I googled “sample rate”, but I don’t understand all the difficult words …
The higher the sample rate, the better the sound quality?

show-frequency-what-is
For those people, here we explain in a way that is easy to understand even for beginners, from the meaning of the sample rate that always appears when starting DTM, the relationship with the sound quality, the confirmation method to the setting method. I would like it in a way.

Please refer to that.

1. What is the sampling frequency?
Adjusting the sample rate Sound quality
Sample rate is a frequency that indicates how accurately sound is captured. .

Sample rate, sample rate Also sometimes called sample rate in English.

The higher the number, the better it will be caught. .

Of course, if you want to import sound to your computer, you have to convert it to data. Capturing sound data is called sampling.

* Currently, sampling mainly refers to capturing existing sounds. (Using existing music, recording material, etc.)

Rate means rate or commission in Japanese.

In other words, sampling = sound data capture. Fee = fee, commission

Literally translated, “sample rate = sound data capture rate. It will be a commission.

In other words, the sample rate indicates how well the sound is captured.

For example, if the sampling frequency is 48 kHz, the data will be taken by dividing it into 48000 times per second.

The higher this value, the finer the sound will be sampled.

It may be easier to understand if you think of the version in which each pixel art block is a sound.

2. Does the sound quality change depending on the sampling frequency?
Sample rate loading sound quality
A common question is whether the sound quality changes based on frequency.

Naturally it changes.

However, beginners and those who do not have expensive equipment will not notice much even if it changes.

So I will show you how to check it!


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What is bit?

What is bit?

Sample rate

bit is an abbreviation for binary digits.
16 bits and 24 bits in catalogs, etc. represent the number of binary digits * handled by computers, etc.

SAMPLE RATE

In digital audio, analog sound is converted to a digital signal,
but the number of bits determines how accurately the amplitude value is converted when it is converted to a binary number (quantization) after sampling.
In the case of 1 bit, only 1 or 0 can be judged, but in 8 bit (10001001), 2 raised to the eighth power, that is, 256 steps can be judged in detail.

Currently, the 16-bit mainstream has 65,536 steps and the 24-bit mainstream has 16,777,216 steps.
Now,
there is a part that does not match the actual waveform (analog waveform) and the quantized and sampled digital waveform. This is called quantization noise.
This noise is especially noticeable when the number of bits is small.

So simply increasing the F’s and the number of bits will improve the sound (closer to the original sound)
, but it will consume a lot of memory. Also, in the case of digital recording, it is
very important to manage the input level to bring out the high quality of the sound.
If the recording level is too low, you won’t be able to bring out its goodness.

It is important to configure it so that it is not clipping at the maximum level of the music to be recorded,
but try to increase the overall average level as much as possible to have a wider dynamic range
(recordable high and low level difference) than analog. Make the most of it and record with a good signal-to-noise (SN) ratio.

* The decimal numbers that we usually use are represented by a combination of 10 types of numbers from 0 to 9, but in
binary numbers, are represented by a combination of 0 and 1.

For example, in a 4-digit binary number,

Decimal number 0 1 2 3 Four ・ ・ ・ ・ 14 15
Binary number 0 1 Ten 11 100 ・ ・ ・ 1110 1111
You can express a number from 0 to 15 as.

(5) What is timing?

It is a state in which each device moves in harmony with each other at the same time in the system.

Digital devices use a reference signal called a word clock, and
Each device can be synchronized with a high precision that cannot be compared with analog devices.

For the configuration of each device, the device that supplies the reference word clock is set as the word clock master, and
all other devices are configured as
word clock slaves so that they can operate synchronously in response to the instruction of a unit set by this master increases.

The role of the word clock is similar to that of the conveyor belt used on factory assembly lines.

The digitized audio data is divided into small times, it is
they are transmitted to each device, they are processed and finally the DA converter returns them to an analog audio signal.
What happens if the speed of the conveyor belt changes along the way?
The data will be lost or the time will not match.

If there are devices in the system that are not synced
, problems such as loss of sound and noise mixing will occur due to the same cause.

With regard to synchronization, if each device is precisely configured and word clock transmission between each device is guaranteed,
can achieve high-performance and comfortable operation unique to digital technology.

Difference between digital and analog

Difference between digital and analog

Audio Sample Rate

The sound is analog. And sound is the vibration of the air. How is this sound vibration transmitted?
For example, when a stone is thrown onto a surface of calm water, the ripples spread around it, but if
Cut in the direction of the waves and look at the cut end, the waveform is as shown

sample rate

Air waves spread from the point where sound is emitted even in air. Although invisible to the eye, it has a
similar waveform. This is the analog waveform of sound.

Therefore, although it is digital, when such a sound waveform is recorded or communicated by phone or wireless, as
shown in Fig. 2, the change in the analog waveform is electrically replaced with a series of numerical values ​​according to a certain promise. ..

When recording or communicating, if you handle it as analog, it is easy for noise to enter and the sound quality to deteriorate, but when trying
the waveform of the sound as digital = numerical data, you can eliminate that worry and
maintain a certain quality. You can do various processing while maintaining it.

(2) What is convenient when it is digital

Digital audio signals are convenient because they can be recorded and edited using a personal computer, for example.

In addition, 74 minutes of music can be recorded on a CD with a diameter of only 12 cm, and through digital compression processing
, music of the same length can be recorded on an MD with a smaller diameter.

Since digital signals can be compressed in this way, it is also convenient for storing large amounts of information.
Not only sound, but also video signals with a higher amount of information can be recorded and communicated at high speed through the use of compression technology.

Especially in communication, a two-way digital multiplex communication can be realized communicating multiple pieces of information with a single wire.
In addition to electrical signals, laser light can also be used for optical communication, making communication possible at extremely high speeds.

(3) What is the sampling frequency?

Digital signals are processed at predetermined fixed time intervals.
The sample rate (sample rate) indicates how many times a second is processed and is expressed as Fs or fs.

The sampling frequency unit is Hz (Hertz), and the
44.1 kHz (kilohertz) sample rate means 44,100 pieces of data are processed per second.
(K represents 1000 times)

AD conversion converts a continuous analog signal into a digital signal,
measures the size of the signal at each moment determined by the sampling frequency (sampling) and converts
the result in a binary number (quantization).

On the other hand, DA conversion converts a digital signal into an analog signal,
It reads the digital signal in the sample rate time interval and connects it smoothly.

Since digital signals can be reproduced up to half the sampling frequency, how much
The higher the sample rate, the higher the playable frequency and the better the sound quality.
In familiar areas, 44.1 kHz is used for CD, and 48 kHz is used for DAT and B modes of satellite transmission.

In addition, recent professional equipment uses high sampling frequencies (high sampling), such as 88.2 kHz and 96 kHz,
and are designed to faithfully reproduce even higher frequency sounds to improve sound quality.

What is a bit rate?

What is a bit rate?

BITRATE

I write “What is a bit rate?”, But most people can say “Bit rate? I have never heard of it”.

bitrate

Have you ever seen “MP3 128kbps” when going to home electronics retail stores?

MP3 is a compression format and the next 128 kbps is the part called the bit rate. This is an indicator of “how much data is converted per second”. The higher the number, the better the sound quality.

Conversely, the lower the number, the higher the compression ratio, but the worse the sound quality.

In other words, what is even more different from CD and MD players is that you can decide the sound quality yourself. (For indecisive people, it means “you have to decide yourself the goodness of the sound quality” (^ o ^) /)

I’m the last person, so it took me two weeks to decide …

If you don’t know, that’s fine. Most media players are set to MP3 128 kbps by default (initial state) or are recommended by the manufacturer’s software. For Windows Media Player, the extension is wma.

So if you don’t mind too much, buy it and import it on your computer and transfer it! It’s okay. Actually, I can’t tell the difference when I set it to 128 kbps or more. Is it a place where you can focus and see the difference between 128 kbps and 192bps? (If you have a good ear, you will be able to understand it in your daily life …)

So when does the bitrate change?
~~ By increasing the bit rate ~~
For example, if you buy a large capacity player, about 10,000 songs will be included. You don’t listen that much, you can’t hear it, right? So if the capacity is full I think it is fine to capture at 192 kbps or the highest 320 kbps. (To be honest, I don’t know the difference between 192 kbps and 320 kbps)

If it is a classical song, you can increase the bit rate by one step. That is a song to enjoy the song. I think pop music can be left as is. I enjoy singing.

It’s okay to change it depending on the song, but it might get annoying soon. .. ..

~~ By reducing the bit rate ~~ Since the
Flash memory player capacity is limited, some people use it with a lower bit rate. However, at 64 kps, you can see that the sound quality is clearly bad. Music like nothing? If so, one way is to downsize it to save capacity.

If you don’t have enough capacity, you can reduce it if you are learning a language. “So should we learn with poor sound quality?” It is true that the sound is a little worse, but the sound range (frequency) is limited to the human voice, so it is better than music. The sound did not get worse. If you don’t need it, 128 kbps is enough.

By the way, I put the difference in the bitrate as shown below. Actually, the file size is compressed to correspond to each one. The optimal compression rate is 128 kbps. Can you tell the difference compared to the original logo?

What is the “clock” on a CD?

What is the “clock” on a CD?

CD Player

The CD player contains a biological clock. You may think it is true, but it is a fact.

Cd Player

A watch is called a “clock” and it actually carries a crystal oscillator (crystal clock) that keeps the exact time. This is not for the timer. Time is important to read the information recorded on the CD, and the crystal clock, which is the body’s clock, plays an important role. Since this is a very high frequency pulse (clock pulse), it splits (slows down the count) and issues the necessary commands to various blocks in the player.

Let’s teach the seeds we are proud of as an ear study. “The clock is related to the pit length of the CD.” In order for the player to read the 0 and 1 information of the hole, it is necessary that the length of the hole and the time of the biological clock coincide exactly, but for that purpose it is not good. The length of the pit is set to an integral multiple of the clock. There are actually only 9 types of wells on the board, from the shortest (3T) to the longest (9T). You can see that T is a clock pulse and it is a well-researched format.

If the clock is wrong, the sound will be cloudy. This is because the pasle’s time axis fluctuates and jitter occurs. Therefore, the topic of discussion among fans is the external clock. If your body clock is poor, there are other, much more accurate cesium and rubidium clocks. You can use this pulse to move the player! This is why some high-end CD players have an external clock input.

Next time, let’s go through the glossary and how to read the optical disc player specifications that have come out so far.

CD Player Sound Quality Enhancement Technology: What are High Bits and High Sampling?

CD Player Sound Quality Enhancement Technology: What are High Bits and High Sampling?

However, CD players have various technologies to improve the sound depending on the manufacturer.

Like Denon’s AL24 processing and Pioneer’s legato link conversion. Even if the name is different for each manufacturer, it basically reproduces the subtle nuances and quirky atmosphere of the original analog audio that was cut on CD using extended technology like high bit and high sample. It’s just a device in CD format, but when you ask it, it certainly feels clear and the amount of information has increased.

So what kind of processing are you doing?

sampling

The left side of the figure is a normal CD format. The horizontal axis is incremented by fs = 44.1 kHz and the sample data is read with 16-bit precision. This is as explained above and unless there is special processing on the player side it will play as is with CD audio.

But the figure on the right is different. This is an image of the AL24 example, and the bits are expanded from the usual 16-bit to 24-bit using a dedicated chip. So a simple calculation can express a fine sound that is 2 to the eighth power, that is, 256 times. It seems that the upper and lower bits are moved and advanced things are done, but due to such bit expansion and high sampling (extending the high frequency range) like 4fs and 8fs in the direction of the horizontal axis, the squares are much smaller . Even if it is a CD, you can enjoy high-quality sound that surpasses that of a CD.

PCM conversion flow

PCM conversion flow

Pulse Code Modulation

Let’s summarize how analog music signals are digitized in PCM and burned to CD. PCM is an abbreviation for pulse code modulation. In Japanese, it translates to pulse code modulation method.

PCM

The music signal is originally a continuous analog signal. A continuous waveform that ripples like a wave will not fit in the hole of a CD as is, so test it first. What part of the rippling wave should be used as a sample? Of course, it is necessary to have regular intervals, and in the case of CD, it is decided to sample at 44.1 kHz. kHz is a unit of frequency and is the number of repetitions per second. We’re going to sample at a tremendous rate of 44,100 times per second. The job of sampling is sampling, and it does not mean that the waves are crushed separately.

After sampling in the direction of the time axis in this way, the next step is how to read the discrete data (points) with what precision. This is the quantification. It’s not used often, but in English it’s called quantizing. Since the vertical axis of the graph is the signal level, that is, the magnitude, the precision point is how many steps to read to the highest point of the wave. The unit is the number of bits.

The bits are a binary number in the digital count. Binary numbers are a game, and as the number of bits increases, the number that can be expressed at an accelerated rate increases (number of steps = sampling precision). The calculation is “2 raised to the power of the bits.” For example, 3 bits would have 2 x 2 x 2 = 8 steps, but 5 bits would have 2 x 2 x 2 x 2 x 2 = 32 steps. It seems that it will be incredible if we continue like this. Yes, 16 bits is 2 to the power of 16, so multiply 2 16 times to get 65536 steps. Remember the “65,000 steps”.

Still, it’s not analog per se, but if you play it on a CD player it will play the original continuous analog wave, which is why digital is Erai. Actually, after quantization, the encoding work is done and a 16-bit PCM digital signal is obtained as “010011 … 10”.

Digital is strict and, in fact, there are some rules. It is often said that “CD has a frequency range of 20 kHz and a dynamic range of 96 dB”. This is determined solely by the format. To put it bluntly, the 20 kHz high-frequency range comes from the sample rate, while the 16-bit quantization defines the D range as 96 dB.

It’s kind of logical, but it’s called “Shannon’s Sampling Theorem (Erai scholar)”, and it can record high frequencies up to almost half the sampling frequency (fs). For quantization, there is a guideline of 6 decibels per bit, which is 6 x 16 = 96 decibels.

What are the sample rate, the number of quantization bits, and the clock?

What are the sample rate, the number of quantization bits, and the clock?

Sample Rate and Bit Depth

There is some format jargon that you really need to know about CDs. It is the “sample rate” and the “quantization bit number”.

Sample Rate and Bit Depth

Related to that, you will deepen your understanding if you also learn about the “clock” from the CD. The next time you learn “How to Read Specifications / Optical Discs”, it will go into your head.

■ What is the sampling frequency and the number of bits?

Digital audio recorded on a CD has a 44.1 kHz sample rate and a 16-bit quantization bit rate, right? Yes, that is correct. It has appeared several times so far, but this is the first time that we have explained it in detail from the basics.

First, let’s start with the image. Just the esoteric feeling of sampling and quantizing, and the “vertical slice” and “horizontal slice” of the signals first. Think of it like cutting a radish. First of all, I’ll cut it vertically with a kitchen knife. You can make a lot of cuts, but they were originally continuous. The solid curve is the analog voice, and the first thing to do when digitizing it is the “vertical slice” = “sample” image.

Next is the quantification work. Even if the cut is a cut, it is quantified to “cross” the kitchen knife on its side. Then the radish will be divided into small squares. Did you imagine that the finer the square, the closer it is to the original analog signal?

The CD format is the rule of how fine the radish is cut (analog signal). “The sampling frequency is 44.1 kHz and the number of quantization bits is 16 bits” means that the first sampling is done at a rate of 44,100 times per second, and then the level is read with an accuracy of 16 bits (2 to power step 16). . Sampling is also called sampling, but in the first place, sampling is the norm, and without sampling, the quantification work cannot be done.

What is the so-called bit rate?

What is the so-called bit rate?

BitRate

A value indicating how many bits of information are processed or sent / received per unit of time.

AUDIO COMPRESSION

Also called transfer fee. The amount of information in one second of audio data and video data is expressed in “bits per second” (bps: bits per second). Usually used in conjunction with “k (kilo)” which represents a unit of thousand or “M (mega)” which represents one million units because the number of digits increases and is expressed as “kbps” or “Mbps” . (1 kbps is 1000 bps, 1 Mbps is 1 million bps). It is often used in the audiovisual (AV) genre, and in the case of audio and image data, the higher the value, the more detailed the information, and the better the sound quality and picture quality. The standard bit rate for MP3, one of the audio compression formats, is 128 kilobits per second (kbps), which compresses uncompressed WAV files (approximately 1400 kbps) with CD sound quality to approximately one-tenth of the amount of information. what are you doing. The video bit rate is higher due to the large amount of information, and the high definition terrestrial digital transmission is about 18 megabits per second (Mbps), and the BS high definition digital transmission is about 24 Mbps. Also There is a unit that expresses the transfer speed, “bytes per second” (Bps or B / s), which is a reference value that expresses the number of bytes per second. Since 1 byte is 8 bits, Bps can be calculated by dividing bps by 8.

Bit rate

It is the data communication speed, which is the amount of data that can be sent and received in a certain period of time. The unit is “bps”, which is short for “bits per second”. It is also used to refer to the amount of data used to express one second of video or audio when compressing video or audio. The greater the amount of data (= lower the compression rate), the more faithful it will be to the original, but a high-speed communication line is required.
On the other hand, as the amount of data is reduced (= the compression rate is higher), the image quality and sound quality deteriorate, but transmission is possible even in an environment where the communication speed is slow .
⇨  bps, transmission.

Processed per unit of time, or the transfer is a bit number. It is generally expressed as a number per second and uses bps as the unit. In a computer network, it is represented by a physical quantity as a communication speed, and in data transfer with a peripheral circuit or device within a computer, it is represented by a physical quantity as a transfer speed. It is also used as a unit to express the amount of information per second when compressing audio and video data, and if this value is the same, the higher the value, the higher the sound quality and picture quality. ◇ Also called “bit rate”, “bit efficiency” and “bit rate”.

What is the best way to use compressed sound sources like MP3, AAC and WMA correctly? Part 2

What is the best way to use compressed sound sources like MP3, AAC and WMA correctly? Part 2

audio compression

User-friendly bit rate of sound quality and capacity is 128 kbps to 160 kbps
The problem is the compression rate (= bit rate) expressed in the unit of “kbps”. Difficult theory aside, it’s okay if you think the point is “bitrate = standard for numerically expressing sound quality”.

audio bit rate

“Reduce the amount of data by reducing the sounds that are not harmful to the human ear” In a compressed sound source, the lower the bit rate, the lower the capacity, but the higher frequencies are cut off. So if you lower the bitrate too much during encoding, you will get some moody sound quality somehow.

・ ~ 96 kbps …… Since the sound does not lengthen, it is suitable for talk-centric radio programs, etc.
・ 128 kbps …… No matter who listens to it, there is not much discomfort. Suitable for pop and rock with PC speakers and car audio
・ 160 kbps …… Sound quality that can be satisfied even with general audio. Suitable for loud jazz
・ 192 kbps …… There are few glitches even when listening with headphones. Even classical music with a wide range is fine.
・ 256kbps / 320kbps …… High sound quality close to that of a CD (1411kbps equivalent)

Although there are individual differences, let’s think about it based on the above. The maximum difference in sound quality that a normal person can hear is 160 kbps. Beyond 192 kbps, you will not notice any difference unless you are a very “hearing” person.

Also, as the number of songs increases to 100 songs and 200 songs, the difference in capacity will be large, so choose a bit rate that is easy to use. If you convert a 4-5 minute song, often found in pop music, to MP3, the capacity will be roughly as follows.

·
128 kbps: Approximately 4 MB · 160kbps: Approximately 5-6MB · 192kbps:
About 7 MB320 kbps
: Approximately 10 MB

AAC and WMA have a higher compression rate than MP3 and the capacity is lower even at the same bit rate. Since it is also resistant to low bit rates, AAC and WMA can sound better at 128 kbps or less.

On the contrary, when it exceeds 160 kbps, MP3 has a superior sound quality in theory. Keep in mind that the higher the bitrate, the better the MP3 will be in terms of sound quality, whether you can listen to it or not.