what do sample rate mean Archives

The difference between 44,100 Hz (music industry) and 48,000 Hz (video industry)

Free Download Mp4Gain

The difference between 44,100 Hz (music industry) and 48,000 Hz (video industry)

44100 vs 48000 sample rate

In video production, record the frame rate for shooting and the sample rate for recording. Remember this is one of the basics for shooting and recording.

SAMPLE RATE

First, about the difference in sampling frequency. Generally speaking

44,100Hz (44.1kHz) is the standard in the music industry

48,000Hz (48kHz) is the sound standard in the video industry

The difference between the two sample rates is just that. I talked about the sample rate as
the frame rate in video in another article, “Sound Principles Required for Video Production,” Sample Rate and Bit Depth. ”
In other words, the higher the sample rate in Hz, the softer the sound will be.

There are several theories about the historical background of 44,100Hz.
I would like to introduce you to one of the most logical.

First, when sampling sound, you need a sample rate that is at least twice the highest frequency you are recording. This is the sample rate necessary to obtain a minimum of the waveform. This is because it is not possible to record a sound that has the character of a wave if there is only one place to take a sample. Most people say that the audible range is 50 Hz to 16,000 Hz. Double is 32 kHz, but it seems that the harmonic components that make up the tone need to be recorded in order to record the voice correctly. Only when this is taken into account does it appear that up to 44,100Hz is required. Click here for more details.

Sound Processing “I want to hear my voice clearly” (link outside of Vook’s site)

What happens when the sample rate is low?
When digitizing analog information, if the sampling rate is not high, the high-frequency information will be hidden in the low-frequency information.
Then the high-frequency sound will be recorded as low-frequency sound.

In any case, by definition, 48,000 Hz has better sound quality than 44,100 Hz. The video industry has introduced 48,000 Hz.

One problem that sometimes occurs is that “I was recording 48 kHz video and the separately recorded microphone was set to 44.1 kHz.” At first I thought that different sample rates would be a big deal, but it doesn’t really seem to be the case.

Sound recorded at a small sample rate just has a small number of samples per second, but since there is almost no difference between 44.1 kHz and 48 kHz, I think you can barely tell the difference when you listen to the sound normally. At 96 kHz, the sound quality is even higher, but the number of samples is so large that ordinary people cannot hear it at all.

In some cases, the sample rate is really important.

1) By writing the audio actually recorded with a different sample number as a video file. This is because the sample rate must be converted to a video sample rate that is different from the conventional 44.1 kHz and 96 kHz sample rates, that is, 48 kHz. Software that specializes in video editing seems to have sound distortion at this point.

2) When recording a 48 kHz music video to match the music played at 44.1 kHz on the site. In this case, it can be very difficult to match performance lips to post-production due to the different sample rates of the sound being played. It’s called sink drift.

3) Another point It seems that this is a problem that occurs at the time of recording, but there is a problem that the sound changes gradually when the sound recorded separately using a cheap recorder is synchronized with the sound recorded in the reference of the video. it seems that there are moments. In this case, it seems that you need to manually fast-forward the video a little and match it to the audio file, or extract a few frames at the important points in the audio and sync it up. It seems that this has nothing to do with the sample rate, so I will describe it so as not to cause misunderstandings.

Free Download Mp4Gain

Mp4Gain Main Window

Mp4Gain Features

Free Download Mp4Gain

What is the sample rate (sample rate)?

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!

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 the sample rate and bit rate?

What is the sample rate?

Sample Rate

Frequency is defined as the number of cycles of periodic motion per unit of time. The SI unit of frequency is called hertz (Hz, after its inventor Heinrich Hertz). One hertz corresponds to one cycle (or complete oscillation) per second.

Sample Rate

Example. Sound waves have a frequency in the range of approximately 20 to 20,000 Hz. This means that at any point along the path of the sound wave, the pressure will fluctuate from high to low, 20 to 20,000 times per second.

In digital audio, the maximum frequency that can be successfully recreated is half the sample rate. Therefore, with a sample rate of 44.1 kHz, frequencies up to 22.05 kHz can be recreated. Wave frequency refers to how many times per second a wave moves from its highest point to its lowest point and vice versa. It is usually measured in hertz (Hz) or cycles per second. The frequency of the wave determines its height. High-frequency waves have a high pitch, while lower frequencies have a lower pitch. The average person can hear frequencies from 15 or 20 Hz to about 20,000 Hz (20 kHz).

Analog wave The wave amplitude refers to half the distance between the highest point of the wave and the lowest point. The greater the amplitude of the wave, the greater its volume, which is generally measured in decibels (dB). The decibel range for human hearing is complex and depends on the frequency of the sound in question, the age of the person and the listening environment, but varies from approximately 0 to 120 dB, with each 10 dB change corresponding to a doubling of the perceived volume.

Absolute Threshold: ATH is the volume level at which a certain sound can be detected 50% of the time.

What is the bit rate?

Bit rate refers to the data transfer rate (that is, how many bits are transmitted in a given time), generally expressed in bits per second. Common units of bit rate are kilobits per second (Kbps) and megabits per second (Mbps). The term is also commonly used when talking about digital sampling and sample rates. For example, the MP3 audio compression algorithm is often configured to output files at a bit rate of 128 kbps. This means that the file contains an average of 128 kilobits for every second of audio (960 KB per minute). This is in contrast to CD audio, which is encoded as 44,100 16-bit stereo samples per second: 1411.2 kbps (16-bit x 44100 Hz x 2ch).

Often times, bytes are written in uppercase and are multipliers (for example, “KB” for kilobytes) and lowercase factors are bits (for example, “kb” for kilobytes). All modern computers use 8-bit bytes.

MP3 bit rate
The MP3 bit rate can be misleading. For example, an MP3 “constant bit rate” (CBR) of 128 kbps will use approximately 128 kilobits for every second of encoded audio (so the file size in bits divided by the length of the audio is approximately 128,000), and Your frame headers will appear at regular intervals, but internally, frame-by-frame, you can encode audio at bit rates higher or lower than 128 kbps by using a bit pool (the ability of a frame to use spare bits from a previous block). However, the size of this bucket, and thus the amount of variability, is limited, so 128 kbps will be very close to the effective bit rate throughout the file.

See also: 8D surround sound and how to do it
As another example, “128 kbps VBR MP3” is often incorrect, as the purpose of VBR is to allow each of the internal MP3 sectors to have its own bit rate. When people refer to the VBR MP3 bit rate, they are generally referring to the actual average bit rate of their frames. If the length of the encoded audio is known, then the “bit rate” can be the data size of the file divided by its duration, which will be fairly close to the same number. However, the length of an MP3 VBR cannot be accurately determined without scanning all the frames.

Audio sample rate and bit depth – in simple, understandable language

Audio sample rate and bit depth – in simple, understandable language

Bit Depth and Sample Rate

What is the sample rate (sample rate)? What is bit depth?

Sample Rate & BitDepth

Even if you are not dealing directly with digital sound recording, you will be interested!

Are you new to the world of digital music? Not sure what all these designations and complex numbers mean?

Hmm, no wonder! After all, every day there is more and more information. And knowing everything is almost impossible.

Yes, this is not necessary! You need to know the essentials.

Sample rate and bit depth are sound engineering concepts that you should know if you decide to make music in a computer environment.

Even if you haven’t had to record music in a virtual environment yet, but have dealt with audio (be it on a portable digital player, a player on a computer, or elsewhere), you may have seen some numbers in the properties of audio: “16 bit, 24 bit, 44100 Hz, 48000 Hz …”

The material is presented briefly and is accessible even to the uninitiated. Just the essentials.

So what are sample rate and bit depth? What is it for?

To begin with, we agreed that in different sources you can find: Sample rate and Sample rate. The abbreviations are equivalent. Call it what you like the most.

And bit and bit depth. It’s the same, the same, it just sounds different.

So.

Sample rate (sample rate) …

All inanimate music (music produced by a computer, music center, etc., that is, not live) has this parameter. This is the number of samples per second. Without going into details, I will say that 44100 Hz is optimal for humans. Since at a higher value, the sounds to be sampled will be practically inaccessible to our ears, we will simply not hear them, because they will be out of earshot.

I’ll explain a bit more in datell about sample rate. Discrete means discontinuous. That is, the sampling process is the processing of each bit of information one by one (that is, discretely and not all at once). In our case, this happens 44100 times per second. By Nyquist’s theorem, the required sampling rate for normal perception should be twice the hearing threshold. Since an average person listens up to 16 KHz (KiloHz or 16000 Hz), and something (normal for a healthy young person) up to 20 KHz, the sampling frequency was determined at 44.1 KHz (44100 Hz), that is, twice the threshold. audibility of the human ear. Why not 40 kHz (40,000 Hz)? Taken with margin (nobody canceled errors and noise on the route and after the CD release).

I hope everything is clear now.

The bitness (Bitness) is a kind of resolution of these same samples. Why am I calling this permission? Just so you prefer to understand by analogy what is what.

Grab your monitor – the higher the resolution, the better the picture, right? At low resolution you will see individual pixels and the eye will no longer be happy as before. I smile

Bitness is dynamic range, that is, the oscillation of your audio up and down (in terms of volume, power, so to speak), the nuances of performance.

The higher the audio bit rate, the more space the audio will occupy on your hard drive (on your computer); keep in mind.

For projects that are important to you, I advise you to use 24 bits and a sample rate of 48000 Hz. THIS IS A STANDARD. Then, for CD output, it will be possible to downgrade the data to 16 bits and 44.1 kHz.

But some people prefer to work on 24/96 (24 Bits – bit depth, 96 KHz – sample rate) or 24 / 88.2. The taste and the color …

For most projects, 16 / 44.1 is adequate (16 bit – bit depth, 44100 Hz is equivalent to 44.1 KHz – sample rate).

The sample rate and bit depth go directly next to each other and never go alone. That is their destiny.

Why is 44,100 used as the high quality sample rate?

Why is 44,100 used as the high quality sample rate?

Sample Rate

Why did we choose 44.1 kHz as the recording sample rate?

Sample Rates

People’s ears hear a sound whose frequency varies between 20 Hz and 20 kHz. By Nyquist’s theorem, the recording speed must be at least 40 kHz. Is this the reason for choosing 44.1 kHz?

Explain in more detail, the sample rate means how many “frames” should be recorded per second to have high quality audio.
According to the famous theorem created by a famous scientist named Nyquist, the sampling frequency must be at least twice the maximum frequency that we will record … then, as the human ear can hear approximately 20 kHz at most, twice that would be 40,000 per which was proposed 44,100 as a standard sampling frequency for high fidelity audio.

It is true that, like any convention, the choice of 44.1 kHz is something of a historical accident. There are several other historical reasons.

Of course, the sample rate must be higher than 40 kHz if you want high-quality audio with a 20 kHz bandwidth.

How to make 48.0 kHz was discussed (this matched well with 24fps and supposedly 30fps movies on North American television), but given the physical size of 120mm, there was a limit to the amount of CD data that could be stored and what an error detection and correction scheme is needed that requires some data redundancy, the amount of logical data that a CD can store (about 700MB) is about half of the physical data. With all of this in mind, at 48 kHz, we were told that it cannot hold all of Beethoven’s 9s, but that it can hold all of 9 on one record at a slightly slower speed. So 48 kHz is not.

However, why 44.1 and not 44.0 or 45.0 kHz or some nice round number?

Then in the late 1970s, there was a product called the Sony F1, designed to record digital audio onto readily available videotape (Betamax, not VHS). It was at 44.1 kHz (or more precisely 44.056 kHz). Thus, it will facilitate the transfer of recordings without oversampling and interpolation from F1 to CD or in the other direction.

My understanding of how this turns out is that the horizontal scan speed of the NTSC TV was 15,750 kHz and 44.1 kHz is exactly 2.8 times. I’m not entirely sure, but I think this means you can have three pairs of stereo samples per horizontal line, and for every 5 lines where you would normally have 15 samples, there are 14 samples plus an extra sample for some checking. for parity or redundancy in F1. 14 samples for 5 lines is the same as 2.8 samples per horizontal line and 15,750 lines per second, which is 44,100 samples per second.

With the transition to digital formats, audio was stored in the form of pseudo-video, which could be viewed as black or white (representing a binary format).

The frequency and field structure used by the television standard is as follows for 60 Hz video: 245 lines per field (excluding the first 35 skipped lines). With three samples per line, that is 60 x 245 x 3 = 44100 = 44.1 kHz.

This convention was later used for the CD format due to hardware compatibility issues (the first computer used to make master CDs used for CD replication was video-based).

Now, with the advent of color television, they’ve had to slow the horizontal line speed a bit to 15,734 lines per second. This setting results in 44,056 samples per second on the Sony F1.

Digital Sound and Sample Rate

Digital Sound and Sample Rate

Sample Rate

Given the wide availability of inexpensive digital audio equipment, we invite you to take a closer look at digital audio.

Sample Rate

Acoustic sound is a continuous process in time and in amplitude, that is, the air pressure changes smoothly with time and does not jump from one value to another. Acoustic sound can be converted into an electrical signal using a microphone that, depending on the change in air pressure, changes the electrical voltage it generates at the output. After the conversion of an acoustic sound into an electrical signal, continuity is maintained in time and in amplitude: the signal voltage changes in the same way that the air pressure changes, which is why this sound is called analog. We can record an electrical signal on magnetic tape and convert it back to sound using a loudspeaker that functions as a “reverse microphone”: it moves air in response to changes in voltage. Respectively,

Despite the fact that the analog electrical signal has regularly served humanity for decades, over time some of its representatives (of humanity) became clear that the analog signal and magnetic recording are not the best ways to transmit and store audio information, since both during transmission and during storage occur. unavoidable losses, i.e. sound degradation. At the same time, the transmission and storage of data on computers that operate exclusively on digital data can be done without any loss. The only question is how to convert analog audio to digital and vice versa.

To solve the first problem, there are special devices known as analog-to-digital converters (ADCs). These devices are capable of converting a continuous analog signal into a sequence of separate numbers, that is, making it discrete (English discrete – separate, consisting of separate parts). The conversion takes place as follows: the device measures the amplitude of the analog signal many times per second and outputs the measurement results in the form of numbers.

Analog signal
Sampling
Sampled signal
As seen in the figure, the measurement result is not an exact analog of a continuous electrical signal. How much does digital sound compare to analog? Obviously, this correspondence will be more complete the more often the measurements are made and the more accurate they are. The frequency at which measurements are taken is called the sample rate. And the precision of amplitude measurements is indicated by the number of bits used to represent the measurement result. This parameter is called the bit depth.

Sampling rate
So, the conversion of an analog signal to digital consists of two stages: sampling in time and quantization in amplitude. Time sampling means that the signal is represented by a number of its samples (samples) taken at regular intervals. For example, when we say that the sample rate is 44.1 kHz, it means that the signal is measured 44,100 times per second (in MO, the more intelligible term “sample rate” is usually used, however, “sample rate “is more correct.).

The main issue in the first stage of converting an analog to digital signal (digitizing) is to choose the sampling frequency of the analog signal. As already mentioned, the higher the frequency, the closer the digital signal is to the analog. However, in proportion to the increase in frequency, the following increases: a) the intensity of the digital data stream and the bandwidth capabilities of the interfaces are not unlimited, especially if several channels are recorded / played simultaneously; b) the computational load of digital effects processors and their computational capabilities are also limited; c) the amount of memory required to store the digital signal. Obviously a compromise is needed.

The choice of the sampling frequency affects the frequency range of the received digital sound or the maximum frequency of an analog signal, correctly represented in digital. The range of frequencies a person hears is believed to be 20 to 20,000 Hz. According to the well-known Nyquist theorem, in order for an analog (continuous in time) signal to be accurately reconstructed from its samples, the sampling frequency it must be at least twice the maximum audio frequency. An audio frequency equal to half the sampling frequency is called the Nyquist frequency and is the maximum frequency that a given digital system can store and reproduce correctly. Thus, if the real analog signal that we are going to digitize contains frequency components from 0 Hz to 20 kHz.

Sampling frequency.

Sampling frequency.

Sample Rate

What is its importance for sound recording?

Sample Rate

Time sampling is a process that is directly related to the conversion of an analog signal to digital. Along with it, the data is quantized in amplitude. Time sampling means measuring a signal at the time of its entire transmission.

A sample is taken as a unit. If in words this is not entirely clear, then in an example it seems more convincing. Let’s say the sample rate is 44100 Hz, the same as that used on audio CDs.

This means that the signal is measured 44100 times in one second.

An analog signal is always higher in saturation than a digital one. And its transformation is an inevitable loss of quality.

The sample rate serves as a kind of benchmark: the higher it is, the closer the digital sound quality is to analog. This is clearly visible in the list below. Shows which sound frequency is best.

As you study it, you will see a direct relationship between sampling and track quality:

1,8000 Hz. This frequency is typical for telephone conversations and voice recording on a dictaphone with a simple set of functions. It is used in audio converted through the Nellymoser codec.
2. 22050 Hz is used in broadcasting.
3.44100Hz. As mentioned above, this frequency is typical for audio CDs, and this figure has long been identified with the highest quality level. And today the format does not lose its positions.
4.48000 Hz. These are the DAT and DVD formats, which have replaced AUDIO.
5.16000 – DVD-Audio MLP-5.1.
6.2822 400HZ is a high-tech Super Audio SACD format.
Also read 3D Builder Windows 10 what it is
The list clearly indicates which sound frequency is the best. In addition, technologies do not stop and new formats appear.

But before making far-reaching plans, a very significant nuance must be taken into account.

Its essence is simple: the higher the sampling frequency, the more difficult it is to achieve it technologically. This requires:

Provide high intensity transmission of digital streams. And this is not possible on all interfaces. And the more channels are involved in the recording (which is typical for musical ensembles), the more complicated the process will be;
be armed with a processor capable of powerful computing operations. But even with the most advanced examples, the possibilities for ultra-high quality sound are limited;
Use it to record computer equipment with a large amount of RAM.
Considering the above information, it is not surprising that the sound frequency equal to 44100 Hz is still the most in demand today.

It has been meeting even the most demanding quality requirements for decades, and at the same time there are all the technical possibilities to achieve it. This last factor is decisive for both normal users and most recording studios.

Even knowing what the best sound frequency is, to achieve this, it is necessary to take care of the technical equipment.

What is the sample rate and how does it help improve the quality of the audio or video?

What is the sample rate and how does it help improve the quality of the audio or video?

It is important to distinguish what is a sample, as opposed to what is analog audio.
When digitizing the music, a digital equipment takes an X amount of “samples”, saves the values of each one of them and thus it will be able to “reconstruct” a sound (a video too).

Sample Rate

As sound and video contain much information, it is necessary to take many samples, in order to obtain as much information as possible to later reconstruct one signal, very similar to the original.

Sample rate

There is a theorem that explains why the number of 44 thousand 100 samples per second was reached, but we will not enter from such a technical point of view.

What is important for you to know is that the minimum for HD quality audio to be considered is 44100 samples per second.

With less it will sound like talking on a landline phone or even talking on a walkie talkie.

With 44100 samples per second, the sinusoidal wave can be reconstructed without the existence of “closing teeth” in the wave, rather it will be possible to obtain a very detailed curvature, without peaks or ridges, without areas with squares.

Some use 48,000 samples per second, already reaching very high levels of audio quality. Of course, the greater the number of samples per second, the greater the use of disk space, whether you use an mp3, aac, flac, etc. But nowadays with large storage disks, that is not a problem, because these formats continue to be small.

If you manage to combine a sample rate of at least 44100 and preferably 48000 and a bitrate of more than 160 kbs, your music will sound very good, really good.

What will be good is that you buy headphones or speakers that are capable of delivering a good quality of audio, as well as the device that will compute this audio. Be it a computer, a player, an ipod, etc.

And obviously, starting from an “original” good. That is, get original audio that has a good quality.

By following these simple steps, without having to go into very technical details, you will have a very good sound quality.

Obviously Mp4Gain is the perfect software to mormalize the volume and even give other touches or tweaks like correcting the equalization, etc.