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Sample rate, where it comes from

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Sample rate, where it comes from

Sample rate

Where does the sample rate for CD-audio 44100 hertz come from?

Sample rate

The standard sample rate for CD-audio is 44100 Hertz. Where and why were these 44100s originally chosen for CD audio production?

Starting from the condition (see Nyquist-Shannon-Kotelnikov) of reproduction of the upper limit of the spectrum at 20 kHz, the sampling frequency should have been chosen above 40 kHz. But at the time of the creation of these standards and the development of CD-DA technology (the second half of the 70s of the last century), there was no generally accepted medium in which to record, edit and store digital sound. And for this, it was decided to use standard VCRs, which in those days worked in U-matic format. The digital signal was encoded by a special encoder into a black and white video pseudo-signal and recorded on a video cassette. The structure of the digital signal had to be linked to the frequency and structure of the fields of the television signal used for recording.

This decision was complicated by the fact that different video recording standards are used in Europe and the US: 525 lines at 60 Hz and 625 lines at 50 Hz, while not all lines can be used to record information. The selected frequency should fit the structure of both video signals. 44100 Hz meet this requirement.

In a 60 Hz NTSC video signal, 35 lines are not used for recording, leaving 490 active lines per frame, or 245 in the field for digital audio recording. When writing three samples to a string, the sample rate will be:

60 × 245 × 3 = 44100.

In a 50Hz PAL signal, 37 lines are not used, leaving 588 active lines per frame, or 249 per field, so the frequency will be:

50 × 249 × 3 = 44100.

Although digital sound at that time had nothing to do with the video signal, video equipment was used in the production of the CD, which determined the choice of sampling frequency.

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Basics of digital sound theory Part 4

Basics of digital sound theory Part 4

Sample Rate

The MP3 algorithm allows you to compress the sound 20 to 30 times while maintaining good quality.

Sample Rate

The full quality of the CD is believed to be preserved at a bit rate of approximately 160 Kbps (the concepts of “sample rate” and “sample bit depth” do not apply to MP3 files). However, in most cases, much more compressed audio is quite acceptable. Therefore, in Flash animations, MP3 compression is usually used, which gives a bit rate of the order of 16-32 Kbps. The Flash player supports a range of bit rates ranging from 16 to 160 Kbps. You must select the most suitable based on film size and sound quality requirements. It is often worth leaving the MP3 file at the same quality as imported (therefore, the Use imported mp3 quality setting is on by default). If the quality changes, then the change should be in the direction of decreasing quality, but not increasing.

If the sound is processed in an external editor, you can take into account the fact that the Flash player supports not only the MP3 algorithm, which is part of the MPEG1 Layer 3 standard, but also newer algorithms (MPEG2 and MPEG2.5), that provide better sound quality when bit depth is low. In addition, the player supports MP3 encoding with both constant and variable bit depth (in the latter case, the best compression ratio is achieved).

The MP3 format is optimal for rash projects. Therefore, in practice, it is practically only used. Furthermore, MP3 files can be loaded dynamically, and they also have very useful ID3 tags with information about this sound.

• Nellymoser. A relatively new compression algorithm developed by Nellymoser Inc. Designed to compress human speech. His main idea is that a human voice can include vibrations with frequencies in a fairly narrow range. The upper and lower components can be discarded. Very low amplitude harmonics are also eliminated. The result is compression comparable to MP3 compression, but the sound quality is higher. More details about the Nellymoser algorithm can be found on the developer’s website http://www.nellymoser.com/.

The Nellymoser algorithm codec is included in the player only in Flash MX.

In the Flash IDE, Nellymoser compression is called Speech. You can adjust the quality / size ratio when using Nellymoser compression by changing the sample rate.

You can also include uncompressed audio in your SWF movie. In the development environment, this mode is called Raw. In this case, you can change the bit depth and sample rate. In theory, you can use uncompressed audio if sound quality is significantly more important than movie size (or, even less likely, if you need to save computing resources). In practice, however, it is better to use MP3 compression with a high bit rate (more than 120 Kbps).

Storage formats
There are quite a few audio formats. By default, Flash only allows you to import two of them.

• WAV. The main format for storing uncompressed audio on the Windows platform. Supports mono and stereo audio, various samples, and bit depths. Usually it is WAV where the analog signal is digitized, and only then is one of the compression algorithms applied. WAV files are extremely large, which is why this format has been significantly replaced by MP3. However, WAV is still the main format for professional sound editors like SoundForge.

• MP3. Audio format using the compression algorithm described above. The main format in the case of Flash, as it perfectly combines good sound quality and a small file size. Also, sound files in this format, unlike WAV files, can be dynamically loaded into a movie using the loadSound () method of the Sound class.

If you have QuickTime 4 or higher installed, you can import files in AIFF, QuickTime, Sun AU formats additionally.

Basics of digital sound theory Part 3

Basics of digital sound theory Part 3

Sample Rate

Compression algorithms

Sample Rate

Let’s try to calculate how much disk space an average CD-quality digitized music composition will occupy. Obviously, for this it is necessary to use the formula t KBF size ⋅ ⋅ ⋅ = where F is the sampling frequency, B is the sample capacity, K is the number of strings, t is the time.

Assuming 44.1 kHz herbal, B = 2 bytes, K = 2 channels, and t = 300 seconds, we get that the digitized song will occupy approximately 50MB.

This means that only about 10 uncompressed songs can be burned to CD. Since every second of digitized CD quality sound takes up almost 200 Kb, this sound will be very problematic to use on telephony, radio or the Internet. Even if you digitize the sound as a single channel with a sample rate of 11.05 kHz and a bit depth of 8 bits, each second will occupy 11 KB.

For ordinary telephone networks, this is too much for sound to be transmitted in a continuous stream. A problem arises: somehow it is necessary to reduce the size of the sound files.

It is solved quite effectively by using various compression algorithms.
Flash Player supports the following types of compression.

• ADPCM (Adaptive Differential Pulse Code Modulation – Adaptive Difference Pulse Code Modulation). This type of compression is based on two ideas. First, it was found that in the vast majority of sounds we perceive, slowly changing low-frequency components prevail. From this fact it follows that the difference between adjacent samples is often small (or rather, significantly less than the absolute value of the samples themselves).

This means that the digitized audio signal can be represented not by the samples themselves, but by the differences between them, which are smaller in magnitude and therefore require fewer bits for description. Second, the coding of the difference between adjacent samples is done taking into account the magnitude of the amplitude and frequency composition, since the human ear has sensitivity limits (the so-called adaptation).

The ADPCM algorithm is actively used in IP telephony. It is poorly suited for streaming music due to the significant distortions it introduces into sound (distortions, of course, get into speech, but are hardly noticeable in speech). The compression ratio for ADPCM is typically low, ranging from 8: 1 to 3: 1. The ADPCM Flash Player codec allows 2, 3.4, or 5 bits to represent the difference between samples. Actually, you can achieve acceptable sound quality with a bit rate (bit rate, that is, the “weight” of a second of sound) of 16 Kbit.

The ADPCM algorithm is significantly inferior to MP3, so it is not worth using such compression in principle. MP3 compression will provide an order of magnitude better quality with the same bit depth. The presence of the corresponding codec is explained by the principles of backward compatibility: the MP3 codec is built into the player only in Flash 4. Before that, only the ADPCM codec was used, which is probably due to the free distribution of this algorithm. The reason ADPCM is still used in IP telephony is that it does not require as extensive math calculations as MP3, so compression can be done on the fly.

• MP3. One of the first and most common compression algorithms based on the so-called psychoacoustic compression. It uses the following characteristics of the human ear:

or if a soft sound follows a very strong one, then we don’t hear it. Therefore, it can be discarded;

or a sound component with a large amplitude masks components close to it in frequency, but with smaller amplitudes. Therefore, they can be slaughtered without noticeable loss of quality;

or the ear’s sensitivity to frequency distortion is low, therefore, if the components are close, they can be considered the same;

o We misperceive very low and very loud sounds, so fewer bits can be allocated for their encoding than for sounds with an average frequency.

Technically, the MP3 algorithm is implemented as follows. The sound is divided into chunks of a certain length called frames, and a forward Fourier transform is applied to each set of samples. Its result is the decomposition of a sound wave into elementary sinusoids of different frequencies: harmonics. The harmonic coefficient determines its contribution to the resulting wave. Harmonic coefficients are compared and the least significant are discarded.

Basics of Digital Sound Theory Part 2

Basics of Digital Sound Theory Part 2

Sample rate

A sample rate of 44.1 kHz is not always ideal.

Samplerate

When transmitting data over a low-bandwidth network, the quality of the sound must be sacrificed in favor of its size, in practice sampling frequencies two, four and eight times lower than 44.1 kHz are usually used:

• 22.05 kHz: the so-called radio quality. Used when encoding the sound of FM radio stations. In the case of Flash, it is good for creating background music and event sounds. For the transmission of a human voice, it is even somewhat redundant;

• 11.025 kHz – telephone quality. A sample rate more suitable for the human voice. Used in 1P telephony;

• 5.5 kHz: sound about to lose the information component. This sample rate can be used to transmit low sounds as well as speech (albeit with mediocre quality).

Flash Player supports sample rates 44.1: 22.05; 11,025; 5.5 kHz. The choice of frequency should be determined by the type of sound, as well as the importance of maintaining the size of the SWF file. However, it should be remembered that there is no point in increasing the sample rate of the audio fragment compared to the initial one. This will not increase the quality, but will only unnecessarily increase the size of the movie.

Bit depth of samples
Bit depth determines how many different amplitude values can be captured during digitizing. If the bit width is 4 bits, then the range of the amplitude value from zero to the maximum will be divided into only 16 bins. Naturally, the error when rebuilding the analog signal will be very high. This bit depth is suitable for representing very simple sounds as well as speech (its quality will be low).

The 8-bit width allows 256 amplitude values to be represented. This is the bit depth used by FM radio stations. It is enough to present any sound in satisfactory quality. 16-bit encoding is optimal. At the same time, it can work with 65,536 amplitude options, which is enough to cover the entire audible range.

The 16-bit format is used for CD recording. Higher quality quantization is only justified in the case of studio sound processing.

Flash Player supports 8-bit and 16-bit quantization for uncompressed formats (for example, WAV) and only 16-bit for compressed formats (MP3 belongs to them). Keep this in mind when importing a sound file into a movie.

Number of channels The
Stereo sound is designed to give the playback sound a natural dimension. This is achieved due to the fact that a different component of sound is reproduced from each speaker. In general, the sound of each channel is a separate sound file, so the size of the stereo sound is proportional to the number of channels supported.

Conventional non-professional sound cards work with two-channel audio. The Flash player also supports the same number of channels. With ActionScript, you can mix the sound of the channels by playing the left channel on the right speaker and the right channel on the left. How this is done, we will talk a bit below.

If the sound is encoded in MP3 format, you can choose one of three stereo formats.

• Dual channel. Each channel receives half of the stream and is separately encoded as mono. It is mainly recommended in cases where different channels contain a fundamentally different signal, for example text in different languages.

• Stereo. The channels are scrambled separately, but the scrambler program can give one channel more space than the other if necessary. Most standard format.

• Joint stereo. The stereo signal is divided into two new channels. One is the average of the original channels and the other is the difference between the channels. In this mode, the sound quality is obtained more frequently than in others.

Unfortunately, in the Flash development environment, you cannot specify which stereo format is used. Therefore, if sound quality is of paramount importance, then the creation of MP3 files with the required parameters should be done using one of the specialized programs.

Basic concepts of digital sound theory

Basic concepts of digital sound theory

Sample Rate

Sound is, in general, the vibrations of an elastic medium.

sample rate

The sound is caused by mechanical vibrations of some object (this can be a string, vocal cords, etc.) in contact with the environment. The frequency of vibration (measured in Hertz) determines the pitch. The higher the frequency, the louder the sound. The human ear can perceive sound vibrations from the air with a frequency of 20 Hz to 20 kHz. The ear perceives the amplitude of the vibration as volume. The higher the amplitude, the louder the sound.

Electromagnetic waves are a direct analog of sound waves. The latter are less susceptible to dispersal by the environment, the information they carry is easier to store and process. Electromagnetic waves are the most important secondary carrier of sound. The transformation of acoustic waves into electromagnetic waves (as well as the reverse operation) is carried out due to the usual induction effect, which consists in the appearance of a current in a conductor when it is placed in an alternating magnetic field.

Simply put, the oscillation of the loudspeaker membrane magnet near the coil induces an alternating current in it. If this current is applied to another speaker, then the magnet on its membrane will move, creating a corresponding sound.

This is how the telephone and the radio work.

Sound converted to electromagnetic waves can be easily stored. For this, some parameter of the carrier must be compared (the depth of the plate track or the degree of magnetization of the film) with the amplitude of the oscillations (that is, the strength of the induced current in the speaker coil) . Sound converted directly to electromagnetic waves is called analog sound. Its main characteristic is the direct correspondence of the electromagnetic waves transmitted or recorded with the acoustic ones.

Digital sound is relatively new. Its main difference from analog is discretion. When digitizing, a special device, an analog-to-digital converter (ADC), measures at regular intervals (approximately 0.001-0.0001 seconds) the magnitude of the amplitude of an electromagnetic wave corresponding to an analog sound form and writes its value to a file with a specified precision. This value is generally called sample, or in jargon, sample (of the sample in English, sample). The same digitization is often called sampling or sampling.

By converting sound from digital to analog (this is done by a device called a digital-to-analog converter (DAC)).

The interpolation (approximation) of the intermediate values of the amplitude is carried out according to the known ones. Since the sampling frequency is usually high, this operation allows you to fairly accurately reconstruct the original analog signal.

The digital form of sound is characterized by five parameters.

1. The sampling rate;
2. Bit size of the samples.
3. The number of channels or tracks.
4. Compression / decompression algorithm (codec).
5. Storage format.

Since each of these parameters is quite specific, we will consider them separately.

Sampling rate
The sample rate determines how many samples per second will be taken when digitizing. If we compare digital sound with digital images, then the sample rate will correspond to the resolution (a more “realistic” analogy is the frame rate in cinema). The higher the sampling frequency, the better it is possible to reconstruct the analog signal based on the digital form of the sound (more precisely, the higher the sampling frequency, the broader the spectrum of frequencies that can be recorded during digitization).
The famous Nyquist-Kotelnikov theorem states that for the correct reconstruction of an analog signal from its digital recording, it is necessary that the sampling frequency be at least twice the maximum sound frequency.

Since the upper listening limit is 20 kHz, ideally the sample rate should be at least 40 kHz. This is why the standard sampling frequency used for recording CDs is 44.1 kHz (so-called CD quality). However, the sample rate can be higher, but this sound quality is only used by recording studios and especially demanding music lovers.

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.