Analog vs Digital Audio

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Analog vs Digital Audio: Understanding the Differences

Analog Audio: The Old School Sound

Analog audio refers to a sound signal that is continuous and unbroken. It is the old school way of recording sound, and it has been around for a long time. In the early days of audio recording, analog technology was the only option. Record players, cassette tapes, and reel-to-reel tapes were all analog formats that produced a unique sound.

One of the main advantages of analog audio is the warmth and depth of the sound. Analog recordings have a certain character that digital recordings simply can’t match. As author Salman Rushdie once said, “Analog is warm, digital is cold.”

However, analog audio is also subject to degradation and noise. Over time, the signal can deteriorate, resulting in a loss of quality. Analog recordings also tend to have more background noise and hiss than digital recordings.

Digital Audio: The Modern Sound

Digital audio, on the other hand, is a more modern method of recording sound. It involves converting sound waves into a series of numbers that can be stored and manipulated. The digital format has become increasingly popular in recent years, and it is now the standard for most audio recordings.

One of the main advantages of digital audio is its precision and clarity. Digital recordings are much more accurate and can reproduce sound with much greater fidelity than analog recordings. They are also immune to the degradation and noise that can affect analog recordings.

However, some people argue that digital recordings lack the warmth and character of analog recordings. As musician Jack White once said, “Digital sounds like it has a condom on it.”

Analog vs Digital: Which is Better?

So, which is better, analog or digital? The truth is, it depends on who you ask. Some people prefer the warmth and character of analog recordings, while others prefer the precision and clarity of digital recordings.

At the end of the day, the choice between analog and digital comes down to personal preference. Both formats have their advantages and disadvantages, and it ultimately comes down to what kind of sound you prefer.

Conclusion: The Best of Both Worlds

At mp4gain.com, we understand the importance of sound quality. That’s why we’ve developed a powerful audio normalization and conversion software that can work with both analog and digital formats. Our software can help you get the best of both worlds by optimizing your audio for clarity and warmth.

As technology continues to evolve, we can expect to see new and innovative ways of recording and manipulating sound. But no matter what the future holds, we will always be dedicated to providing our customers with the highest quality sound possible.

Final Words:

In the end, whether you prefer analog or digital audio comes down to personal preference. Both formats have their advantages and disadvantages, and it’s up to you to decide which one is best for you. But with the right tools and techniques, you can achieve great sound quality no matter what format you choose.

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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.

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.

Analog Audio and Digital Audio

Analog Audio and Digital Audio

Analog vs Digital Audio

A sound wave is a kind of complex function, the dependence of the amplitude of a sound wave on time.

Analog Audio vs. Digital Audio

The information contained in the acoustic wave is not determined by the parameters of the medium in which the elastic wave propagates, and the oscillation parameters (amplitude and frequency, tone and harmonics).

Any form of recording (mechanical and Skye, magnetic, optical, laser) is based on the previous conversion of the sound wave into an alternating electrical current with the same parameters of the oscillations (via microphone).

Analog sound is represented on the device as a continuous electrical signal.

Sound quality depends on the fidelity of the waveform, which is very difficult to maintain.

Until 1982, the world was consuming “canned music” only from analog media: vinyl records and magnetic tapes.

Good vinyl records, played with good equipment, offered excellent sound quality, which unfortunately deteriorated a little with each listening due to mechanical wear as the stylus moved along the sound groove and into the dust that permeated everything.

Tape recorders required precision read heads and high tape feed speeds to reproduce smoothly. Over time, the tape demagnetized, the magnetic layer crumbled.

But the main disadvantage of analog audio recording is the inevitable loss of quality when copying.

The mystery of trigonometry

According to the theory of the mathematician Jean Baptiste Fourier, a sound wave can be represented as a spectrum of frequencies included in it.

The frequency components of the spectrum are sinusoidal oscillations (pure tones), each of which has its own amplitude and frequency.

According to Kotelnikov’s theorem, any vibration, even the most complex shape (for example, a human voice), can be recovered unambiguously and without loss from its discrete samples taken with a frequency equal to its doubled maximum frequency.

Vladimir Aleksandrovich Kotelnikov (1908-2005) – a prominent Soviet and Russian scientist in the field of radio engineering, radiocommunication and radio astronomy.

Observation . The finite duration signal has an infinitely wide spectrum. Therefore, when a signal with a finite duration is sampled, it is impossible to recover it from the samples without loss of quality.

Digitization of audio information

The digitization of sound is the recording of the amplitude of the signal at certain intervals and the recording of the amplitude values obtained in the form of rounded digital values.
Any computer includes a motherboard, an audio adapter (sound card).

Sound cards include: ADC (analog to digital converter), synthesizer, mixer, DAC (digital to analog converter) amplifier s, MIDI interface port for gaming devices.

To record digital sound, the ADC produces:

temporal sampling of a continuous signal (determines the value of the amplitude of the signal with the frequency necessary to recreate its original shape = twice the maximum frequency of the sound wave);

quantization by the levels of the measured signal values (determines the number of fixed values (levels, gradations) of the amplitude of the signal);

signal coding (writing in a binary number system).

The reverse operation is performed by the DAC (digital to analog converter).

Bitrate

Bit rate (bit rate): literally bits of information of the transmission rate.

The bit rate is the effective information transmission rate through the channel (the transmission rate of “useful information”, in addition to the service information) expressed in kilobits per second (kilobits per second, kbps).

In lossy compression video and audio transmission formats, the bit rate parameter expresses the degree of compression of the stream and thus determines the size of the channel for which the data stream is compressed.

P-mode compression data stream:

with constant bit rate (constant bit rate, CBR) – The required bit rate is initially set, which does not change throughout the file. It makes it possible to predict the final file size quite accurately, but it does not provide an optimal size / quality ratio for musical works, the sound of which changes dynamically over time.

with variable bit rate (VBR): the codec changes the value of the bit rate based on the desired quality level according to the psychoacoustic model. It offers the best quality of the output file, but its size is unpredictable (it may differ several times).

with an average bit rate (ABR): a hybrid of constant and variable bit rates: the user sets the bit rate in kbit / s and the program varies it within certain limits.