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MIDI and digital sound: pros and cons

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MIDI and digital sound: pros and cons

Digital Audio

The WAVE format is one of many, but it is far from the only format for recording digital audio.

Digital Audio

Unlike MIDI data, digital audio data is actually sound recorded in thousands of units called samples. Digital data represents the amplitude (or volume) of a sound at discrete moments. The sound of digital data is independent of the playback device and therefore always sounds the same. But you have to pay for this with large volumes of sound files.

MIDI data is to digital data what vector graphics are to bitmaps. In other words, MIDI data depends on the audio playback devices and digital data is independent. Just as the appearance of vector graphics depends on the printer or monitor screen, the sound of MIDI files depends on the MIDI device to play these files. Similarly, a melody played on a concert piano will sound different from a normal piano. Digital data, on the other hand, is identical and independent of the reproduction system. The MIDI standard is similar in this respect to the PostScript standard and allows you to control instruments in understandable language.

Compared to digital sound, MIDI has the following advantages:

MIDI files take up less memory and the size of these files does not affect sound quality. On average, MIDI files are 200 to 1000 times smaller than digital files and therefore take up a small amount of RAM, disk space and do not require large CPU resources.

In some cases, MIDI files sound better than digital audio files. In this case, the sound source of the MIDI files must be of high quality.

You can change the length of MIDI files by changing the tempo of the sound, while maintaining the quality and volume of the sound. MIDI data can be easily edited, even at the single note level. You can manipulate small segments of a MIDI song (with millisecond precision), which is not possible with digital audio.

The main disadvantage of a MIDI file comes from its merits. Since MIDI data is not sound itself, playback will be as accurate as the device for playing the MIDI data is identical to the device used to create the original file. Even the sound of a MIDI instrument according to the General MIDI standard depends on the electronic playback device and the method used. MIDI sound is not used for voice playback.

The main advantage of digital audio over MIDI sound is that the reproduction quality of digital sound is always constant, and here MIDI sound is inferior to digital sound. There are two reasons why you should work with digital audio:

A wider selection of programs and systems that support working with digital sound.

The preparation and creation of digital sound elements does not require knowledge of music theory, which cannot be said for MIDI data.

Sound tips
Voice recording from microphone
Any book devoted to multimedia necessarily contains a section on microphone sound recording. In addition, the Sound Recorder (Phonograph) program, which is included in the standard Windows distribution, is usually used for this. Working with it is described in detail in the attached help file. It is easy to use and we will not dwell on it in detail.

The microphones come in condenser and dynamic microphones. Capacitors are more expensive, they give better sound, but your connection must be compatible with a sound card. And the vast majority of sound cards are designed for dynamic microphones.

Another important characteristic of a microphone is its directivity. The microphones are omni-directional (they have the same sensitivity to sound in all directions), unidirectional (they have the highest sensitivity to sound coming from the front), and bi-directional (more sensitive to sound coming from the front and rear). A unidirectional microphone is usually the best option, as it eliminates background noise. But it is more expensive than omni-directional microphones and is more sensitive to choppy breath sounds.

Be sure to pay attention to the impedance (impedance) of the microphone. The optimal value is around 600 ohms.

Therefore, we recommend a 600 ohm dynamic omni-directional microphone.

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Differences between analog and digital audio

Differences between analog and digital audio

Analog vs Digital

Sound information. Sound is a wave that travels through air, water, or other medium with a continuously changing intensity and frequency.

Digital vs. Analog

A person perceives sound waves (air vibrations) with the help of hearing in the form of sound of varying volume and pitch. The greater the intensity of the sound wave, the louder the sound, the higher the frequency of the wave, the higher the pitch of the sound (Fig. 1.1).

Dependence of the volume and pitch of the sound on the intensity and frequency of the sound wave.

The human ear perceives sound at a frequency of 20 vibrations per second (low sound) to 20,000 vibrations per second (high sound).

A person can perceive sound in a wide range of intensities, in which the maximum intensity is 1014 times greater than the minimum (one hundred thousand billion times). A special unit of “decibels” (dbl) is used to measure the volume of sound (Table 5.1). Decreasing or increasing the sound volume by 10 dB corresponds to a decrease or increase in sound intensity by 10 times.

Provisional discretization sound. In order for a computer to process sound, a continuous audio signal must be converted to a discrete digital form using time sampling. A continuous sound wave is divided into separate small time sections, for each section a certain value of sound intensity is set.

Therefore, the continuous dependence of the loudness of the sound at time A (t) is replaced by a discrete sequence of loudness levels. On the graph, this appears to replace a smooth curve with a sequence of “steps” (Fig. 1.2).

Sync Audio Sampling

Sampling frequency. A microphone connected to the sound card is used to record analog sound and convert it to digital format. The quality of the digital sound obtained depends on the number of measurements of the sound volume level per unit of time, that is, the sampling frequency. The more measurements that are made in 1 second (the higher the sampling frequency), the more accurately the “ladder” of the digital audio signal repeats the curve of the dialogue signal.

The audio sample rate is the number of sound volume measurements in one second.

The audio sample rate can vary between 8000 and 48000 sound volume measurements per second.

Audio encoding depth. Each “step” is assigned a specific value for the sound volume level. Loudness levels of sound can be viewed as a set of possible states N, for which a certain amount of information I is required, which is called audio coding depth.

Audio encoding depth is the amount of information required to encode the discrete volume levels of digital audio.

If the encoding depth is known, then the number of digital audio loudness levels can be calculated using the formula N = 2I. Let the audio encoding depth be 16 bit, then the number of sound volume levels is:

N = 2I = 216 = 65536.

During the encoding process, each sound volume level is assigned its own 16-bit binary code, the lowest sound level will correspond to the code 0000000000000000 and the highest – 1111111111111111.

The quality of digitized sound. The higher the sampling frequency and depth of the sound, the better the sound of the digitized sound. The lowest quality of digitized sound, corresponding to the quality of telephone communication, is obtained at a sampling rate of 8000 times per second, a sampling rate of 8 bits, and by recording an audio track (“mono” mode). The highest quality of digitized sound, corresponding to the quality of an audio CD, is achieved with a sampling rate of 48,000 times per second, a sampling rate of 16 bits and the recording of two audio tracks (stereo mode) .

It should be remembered that the higher the quality of the digital sound, the greater the volume of information in the audio file. You can estimate the volume of information in a digital stereo sound file with a duration of 1 second with an average sound quality (16 bits, 24,000 measurements per second). To do this, the encoding depth must be multiplied by the number of measurements in 1 second and multiplied by 2 (stereo sound):

16 bits × 24,000 × 2 = 768,000 bits = 96,000 bytes = 93.75 KB.

Sound editors. Sound editors allow you not only to record and play sound, but also to edit it. Digitized sound is presented in sound editors visually, so copying, moving, and deleting parts of the audio track can be easily performed with the mouse. Also, you can layer tracks

Analog or digital audio?

Analog or digital audio?

Analog vs. Digital Audio

Mechanical, electromechanical, optical, and magnetic recording were originally analog recording methods: recording and reproducing sound vibrations in their natural form (waves).

ANALOG vs. DIGITAL AUDIO

Many people believe that there is no better sound recording than analog. The warm analog sound of the magnetic tape is the standard of the best audio recordings for all mankind. Everyone from Elvis Presley and the Beatles to the latest electronic musicians have used and are using analog tape recording or emulation to create their music.

But analog recording is not the most accurate way to record sound. Rather the most beautiful. Analog sound is pleasant to the human ear due to the presence of “warm” harmonics, which are, in fact, distortions of sound. The most accurate sound recording principle today is digital recording.

The father of digital sound was 25-year-old Volodya Kotelnikov, who created it in 1933. The famous “report theorem” (also known as “Kotelnikov’s theorem” or “Nyquist-Shannon theorem). This theorem was the beginning of the creation of the principle of digitizing sound: encoding an audio signal into bits, that is, converting an analog signal into digital. It only took 49 years to create the CDs we know about. the world, it was only adopted in 1982.

A complete list of the types of digital sound recording in use today is digital magnetic recording (format: DAT cassette), magneto-optical recording (miniDisc format), laser recording (CD, SACD formats), digital recording optical (dolby digital)

The development of computers and digital technology has opened up enormous possibilities for processing and recording sound. Huge analog studios with countless multi-kilogram recording equipment, consoles, and sound processors are being replaced by virtual studios that fit into the computer’s system unit.

To process sound on a computer, it must first be recorded in digital, encoded format. The analog signal is encoded by an analog-to-digital converter (ADC). To play back the recording, you must reverse the digital-to-analog audio conversion using a digital-to-analog converter (DAC). The DAC and ADC are part of the computer sound card and other digital audio equipment. The quality of sound recording and playback is highly dependent on the quality of the DAC and ADC.

DAC and ADC

The main parameters of digital sound are sample rate and bit depth. Both the quality of the digitized sound and the size of the recorded file depend directly on them.

Sampling rate (sampling)

Analog recording begins by pressing the “record” button and ends by pressing the “stop” button. Digital recording is discreet. It consists of many recording fragments (samples) that follow one after another. The number of samples logged per second is the sample rate. It is calculated in hertz. The 44 100 Hz sampling rate (standard for CD) means that the audio signal is measured 44 100 times per second. The lower the sampling frequency, the smaller the frequency spectrum that is recorded. The higher the sampling frequency of the source material, the higher the quality and the larger the file size. When you talk on the phone, you only hear a small mid-range range. This is because the sample rate for phone calls is only 8,000 Hz. To transmit a range of frequencies that the average person’s ear hears and transmits home stereos: 40,000 Hz is sufficient. If the difference in sound quality between 32 and 44.1 kHz is obvious, then the higher the sampling frequency, the less perceptible or not at all perceptible to the ear the difference in quality between the two different frequencies will be. A higher sample rate describes sound more precisely, but at the same time describes those frequencies that the human ear can no longer hear, although changes in sound in the inaudible frequency range can still affect audible frequencies, so that studio recording is performed at a higher sample rate. Since consumer equipment is primarily designed to reproduce sound with a sample rate of 44.1 kHz, when the recording is ready, it is re-encoded to a generally accepted standard. If the difference in sound quality between 32 and 44.1 kHz is obvious, then the higher the sampling frequency, the less perceptible or not at all perceptible to the ear the difference in quality between the two different frequencies will be.

ANALOGUE AND DIGITAL

The essential

Analog and digital are two methods of transporting and storing data. (audio, photo, video …). Analog was born with the onset of electricity, while digital appeared more recently in the computer age.

Analogue and Digital

The analogical principle is to reproduce the signal to be recorded (audio, video …) in a similar way in a medium (magnetic in general). For example, when recording an audio signal in an analog system, the signal present on the tape will follow the same amplitudes (“the same curve”) as the sound wave (with more or less fidelity): the characteristic pressure variations of a sound wave will result in variations of an electrical signal. Therefore, the electrical amplitude of the analog signal will be the more or less faithful image of the signal to be recorded (audio, video, etc.)

analogue vs digital

In digital, the analog signal to record becomes digital thanks to an analog> digital converter. After this conversion, the signal is just a sequence of “0” and “1”, that is, a signal with two amplitudes instead of infinity in analog.

Once in this form, the signal can be copied and transmitted losslessly because instead of transporting a signal whose amplitude should faithfully vary from the original, it carries a signal consisting of only two amplitudes (for example, 0 = 0 volts and 1 = 5 volts). So when a parasite disturbs an analog signal, digitally this parasite will have no effect: for example, a parasite that adds 0.2v disturbance will deteriorate an analog signal, while this same parasite on a digital signal will have no effect because 0v + / – 0.2v will always be considered = “0”.

Therefore, the digital signal is an analog signal made up of two possible levels (for example, “0” = 0v and “1” = 5v) and when the analog signal moves away from these two voltages, it does not matter because all the signals close of 0v will be considered = “0” and any signal close to 5v will be = “1” with a voltage threshold between the two resulting in exceptional immunity against parasites and exemplary ease of making perfect copies (clones) of this signal type.

After digital transport and storage, any signal (video or audio) must return to its original analog form. For example, an audio signal will be converted from digital to analog and then amplified; in fact, our ears cannot hear digitally! ! !

It should be noted that digital is only used (in the case of an audio or video signal) for the transport and storage of data.

2 / Analog VS Digital and data compression.

So we have just seen that digital is not a commercial device, it is the support of all new technologies, but should we consider that digital is definitely better than analog?

I would try to answer yes, but it is necessary to moderate this judgment and that for two reasons: compression and sampling.

* / Digital compression:

Converting a signal to digital is not without problems: the bandwidth occupied by a signal once digitized is greater than its analog equivalent. When dealing with very bulky data, such as video, the processing is very cumbersome; Therefore, it is necessary to use a compression algorithm intended to reduce the amount of information by reducing the quality of the signal. All of the compressions used for video or audio rely on human perception to make this drop in quality little or not noticeable to humans. But the problem is that we often tend to want to compress the signal too much and the degradation becomes noticeable (case of GSM phones, MP3 <128kbit / s, mpeg-1 videos …)

For example, a high-quality audio signal recorded on a high-end cassette deck (analog) will be of better quality than a 112kbit / s mp3, because a large amount of information has been suppressed at 112kbit / s to satisfy the 112kbit / s imposed bit rate.

On the other hand, in an analogous way, the concept of copy generation intervenes: one copy will be less good than the original, one copy of the copy will lose a little more … After 10 or 20 copies of copies, the signal is totally deteriorated and cannot be used. In digital, a copy is often a clone of the original, so in theory you can make countless copies that are strictly the same quality as the original. I am talking about theory because errors can occur during a digital copy, often due to the state of the media. This is the case when copying an audio CD digitally (from your CD-ROM drive to your hard drive, for example): when a scratch or failure occurs, the information will not be copied and will be replaced by the correction system errors due to “extrapolated” data make this defect go unnoticed; In this case, the digital copy will no longer be identical to the original.

Which is better, analog or digital audio?

Which is better, analog or digital audio? Is there really a difference? Do you need very expensive audio equipment to make a difference? Really matters?

analog versus digital

Before we get to the heart of the matter, we should take a quick look at what makes a sound digital or analog. This is how a sound is recorded. A copy of an analog sound recording is a continuous electronic signal.

Today, advances in analog-to-digital conversion methods have improved the quality of digital recordings. Some say that there is no distinction between digital and analog mode. Others disagree, sometimes with passion. Music lovers, those who want the best possible quality in public address systems, insist that analog systems provide better sound.

What are the differences between analog and digital recordings? Read on to find out.

analog vs digital

History of digital sound.

Before the 1970s, music was recorded with analog recording equipment. The microphones they used recorded sound and generated an analog waveform that other devices could transfer directly to the appropriate medium, which was generally a magnetic tape. Assuming the musician wore reliable equipment, the recorded sound was a faithful representation of the original sound.

With digital recording, sound engineers can convert analog waveforms to digital signals. There are many different types of equipment that can be converted from analog to digital. Some studios record analog sound on the original master tape and then transfer it in digital format. Others use special equipment to record digitally directly.

The first digital recordings sacrificed fidelity, or sound quality, in favor of reliability. One of the disadvantages of the analog format is that analog media tends to wear out quickly. Vinyl records can become deformed or scratched, which can significantly affect sound quality. The magnetic tape eventually wears out and is vulnerable to magnets, which can erase or destroy the data stored on the tape. Digital media like CDs can be played indefinitely and are more durable.

Analog versus digital

Some music lovers believe that digital recordings are insufficient when it comes to accurately reproducing sound. They use complex language and jargon to describe the capabilities and flaws of an audio system. Most of his criticisms relate to the frequency of the sound.

Humans can hear sounds ranging from 20 hertz (Hz) to 20 kilohertz (kHz). The frequency of a sound wave corresponds to our perception of the tone of a sound. The higher the frequency, the higher the pitch we hear.

Audiophiles describe the sound quality of an audio system at different frequencies using terms like full, warm, and airy. A full or warm sound comes from a system that reproduces low frequencies well. An aerial sound means that the music played gives the listener the impression that the instruments are in a spacious environment and generally refers to sounds in the high frequency range.

Some music lovers say that vinyl albums are better at low frequencies, which means they provide warm sound. They claim that CDs are not as accurate in reproducing sounds in this range. Others insist that there is no detectable difference between a well-produced digital file and a vinyl in good condition.

If the artist uses an analog format to create the original recording, an analog copy is preferable. In fact, there would be no need to convert sound from analog to digital. The copy must be an exact representation of the original track. But if the artist uses digital recording, it is better to buy the album on CD.

The perception of musical quality is subjective. Two people listening to the same music, with the same equipment, may have different opinions on the quality of the recording.

Differences between analog and digital sound: analog and digital

Analog and digital signals
The sound is of course an analog signal and the analog signal is continuous. Therefore, there is no rest or interruption. Digital signals are not continuous. Specific values are used to represent the information. In the case of sound, a sound wave is represented by a series of values that represent pitch and volume for the duration of the recording.