24/192 digital audio format and why it doesn’t make sense. Part 3


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24/192 digital audio format and why it doesn’t make sense. Part 3

24 bit

192 kHz is considered harmful

24 bit

192 kHz digital music files offer no benefit, but still have some impact. In practice, it turns out that its playback quality is slightly worse, and ultrasonic waves appear during playback.

Both audio converters and power amplifiers are susceptible to distortion, and distortion tends to build up quickly in the high and low frequencies. If the same speaker reproduces the ultrasound along with the frequencies of the audible range, any non-linear characteristics will change part of the ultrasonic range to the audible spectrum in the form of uncontrolled random non-linear distortions that cover the entire range of audible audio. Non-linearity in a power amplifier will have the same effect. These effects are difficult to notice, but testing has confirmed that both types of distortion can be heard.

The graph above shows the distortion resulting from intermodulation of 30 kHz and 33 kHz audio in a theoretical amplifier with a constant harmonic distortion (THD) of approximately 0.09%. Distortion is visible across the spectrum, even at the lowest frequencies.

Inaudible ultrasonic waves contribute to intermodulation distortion in the audible range (light blue area). Systems that are not designed to reproduce ultrasound often have higher levels of distortion, around 20 kHz, which further contributes to intermodulation. Expanding the frequency range to include ultrasound requires compromises that reduce noise and distortion activity within the audible spectrum, but in any case, unnecessary reproduction of the ultrasonic component will degrade reproduction quality.

There are several ways to avoid additional distortion:

An ultrasound-only speaker, amplifier, and signal spectrum splitter to independently separate and reproduce ultrasound you can’t hear so it doesn’t affect other sounds.
Amplifiers and transducers designed to reproduce a wider spectrum of frequencies so that ultrasound does not cause audible harmonic distortion. Due to the additional cost and complexity of the performance, the additional frequency range will reduce the quality of reproduction in the audible spectrum.
Well-designed speakers and amplifiers that do not reproduce any ultrasound.
For starters, you don’t need to encode such a wide frequency range. You cannot (and should not) hear ultrasonic harmonic distortion in the audible frequency band if there is no ultrasonic component.
All of these methods are meant to solve a problem, but only 4 ways make sense.

If you are interested in the capabilities of your own system, the following samples contain: 30 kHz and 33 kHz audio in WAV 24/96 format, a longer FLAC version, some melodies, and a cut of normal songs at 24 kHz to make them drop fully in the ultrasonic range of 24 kHz to 46 kHz.

Tests to measure harmonic distortion:

30 kHz audio + 33 kHz audio (24 bit / 96 kHz) [5 second WAV] [30 second FLAC]
Tunes 26 kHz – 48 kHz (24 bit / 96 kHz) [10 second WAV]
Tunes 26 kHz – 96 kHz (24 bit / 192 kHz) [10 second WAV]
Cutting songs down to 24 kHz (24-bit / 96 kHz WAV) [10-second WAV] (original cut version) (16-bit / 44.1 kHz WAV)
Suppose your system is capable of playing all formats with sample rates of 96 kHz [6]. When playing the files above, you shouldn’t hear anything, no noise, hiss, clicks, or other sounds. If you hear something, then your system has a non-linear response and causes audible non-linear distortion of the ultrasound. Be careful when turning up the volume, if you enter the digital or analog clipping area, even a soft clipping can cause strong intermodulation noise.

In general, it is not a fact that harmonic distortion of ultrasound is audible in a particular system. The distortion introduced can be negligible and quite noticeable. In any case, the ultrasonic component is never a merit, and in many audio systems it will lead to a sharp decrease in the quality of sound reproduction. In systems where it does not damage, the ability to process ultrasound can be preserved or instead, resources can be used to improve the sound quality of the audible range.

Misunderstand the sampling process

Sampling theory is often incomprehensible without the context of signal processing. And it’s no wonder that most people, even brilliant doctors in other fields, don’t get it. It’s also not surprising that many people don’t even realize that they are making a mistake.


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24/192 digital audio format and why it doesn’t make sense. Part 2

24/192 digital audio format and why it doesn’t make sense. Part 2

16 bit vs. 24 bit Audio

Perfect hearing or hereditary gift

24 bit

When I receive many letters, I see that many people believe in the existence of unique people with exceptional hearing. Are there really such people with “golden ears”?

It depends on what you call exceptional hearing.

The healthy ears of young people hear better than the ears of the elderly or damaged ears. Some people are exceptionally well trained to hear all the nuances of sound and music that most people don’t even know exist. In the 90’s, it could recognize all mp3 encoders (they were all pretty bad at the time) and it could prove it in a double-blind test [2].

If a person has healthy ears and is well trained to recognize sounds, I would say that their hearing is exceptional. However, people with below average hearing may be able to notice details that elude inexperienced listeners. Exceptional hearing is largely a matter of training, not the ability to hear beyond the hearing range of ordinary mortals.

Hearing researchers would love to find someone with exceptional hearing and the ability to hear outside the auditory range to test and record the research results. I have nothing against ordinary people, but every scientist wants to find a person with genetic peculiarities to write a first-class article. We haven’t found such people in 100 years of testing, so they probably don’t exist. So sorry. But we will continue to search for more.

Love for the color spectrum

You may be skeptical about everything I just wrote because it goes against all marketing tactics. Instead, suppose people have a craze for color and deviate from the subject of sound.

The figure above shows a rough scale of the sensitivity of rods and cones in the human eye, compared to the visible spectrum. These senses respond to light in overlapping spectral bands, just as the hair cells in the ears are tuned to perceive overlapping sound frequency bands.

The human eye sees a limited range of light waves called visible radiation. Here is a direct analogy with the audibility range of sound waves. Like the ear, the eye has sensitive cells (rods and cones) that capture light in different but overlapping frequency bands.

Visible radiation begins at a frequency of approximately 400 THz (dark red) and extends to 850 THz (dark purple) [3], but visual acuity decreases with the course of life. Outside of this approximate range, the intensity of light entering your eyes can burn your retina. So it turns out that the range is quite decent even for young, healthy and genetically gifted individuals, a range that is analogous to a wide range of the audio spectrum.

Suppose in our hypothetical world, where there is a craze to expand the visible spectrum of video recordings, there is a group of people who believe that these restrictions are not generous enough. They believe that video is not only the visual spectrum, but also infrared and ultraviolet radiation. Continuing with the comparison, let’s assume that the most active part of the group (who is proud of it!) Also claims that this spread spectrum is not enough, and the video will appear more natural if microwaves and X-rays are reached there. For those who have an “eye is a diamond”, the difference will be enormous, just day and night!

Of course, this is ridiculous.

No one can see X-rays (not infrared, not ultraviolet, not microwave). No matter how strongly a person believes in what they can, the retina simply does not have the tools to perceive them.

Here’s an experiment anyone can do: Go and grab the Apple IR Remote [TV]. The LED emits a wavelength of 980 nm, roughly equal to a frequency of 306 THz, which is close to the infrared spectrum. Waves of this length are not that far out of the visible range. Take the remote control to the basement or darkest room with the lights off in your house in the middle of the night and let your eyes get used to the dark.

The image above is an Apple TV infrared remote control, captured with a digital camera. Although the emitter is bright enough and the frequency of the radiation is close to the frequency of the red part of the visible spectrum, infrared radiation is completely invisible to the human eye.

Can you see how the remote control’s LED lights up when you press the [4] button? No? Even a little peek? Try some other remotes, many of them use infrared in the 310-350 THz range.

24/192 digital audio format and why it doesn’t make sense. Part 1

24/192 digital audio format and why it doesn’t make sense. Part 1

Bit Depth

Unfortunately, there is no point in recording music 24/192. Its fidelity does not dramatically exceed 16/44 or 16/48 formats, but it takes up 6 times more space.
Save and read later –

Bit Deph

Earlier headlines reported that musician Neil Young and Apple founder Steve Jobs were discussing a possible launch of a service to download “uncompromising studio quality” music formats. Most of the newspapers, magazines and users were quite optimistic about the prospects of a digital music format with signal quantization in 24 bits, at a sampling frequency of 192 kHz.

Unfortunately, there is no point in recording music 24/192. Its fidelity does not dramatically exceed 16/44 or 16/48 formats, but it takes up 6 times more space.

Today, there are several problems associated with audio quality and the “application” of digital music distribution. The 24/192 format does not resolve any of them. As long as everyone regards this format as a panacea, we will not see any improvement in the field of music.

Let’s start with the bad news

Over the past few weeks, I have talked to smart, scientific people who believe in the 24/192 music format and don’t understand how anyone can disagree with it. They asked good questions that are worth answering in detail.

I also wondered what could be causing such active support for high sample rate digital audio. The responses showed that few people understand the basics of signal theory or the sampling theorem (the Kotelnikov or Nyquist-Shannon theorem), which is not surprising. Misunderstandings about mathematics, technology and physiology were evident in the speeches of many professionals with extensive experience in audio technology. Some have even argued that Kotelnikov’s theorem does not explain how digital audio works [1].

Disinformation and prejudice only play in the hands of charlatans. Let’s go through the basics of why the 24/192 format doesn’t make sense before presenting other more valid ideas.

Gentlemen, welcome! Your ears!

The ear listens with the help of hair cells, which are located on the resonant basilar membrane in the cochlea of ​​the inner ear. Each hair cell is precisely tuned to a specific narrow frequency range, which is determined by the position of the cell on the membrane. The peak of the sensitivity is in the middle of the frequency range, which gradually decreases in both directions and takes an asymmetrical cone-shaped shape, overlapping the frequency ranges of neighboring cells. We do not hear sound if there are no hair cells tuned to that frequency.

The left side of the figure shows a cross section of a human snail with a basilar membrane (beige in color). The membrane is designed to resonate in different places along its length, depending on the incoming frequency: high frequencies resonate closer to the base and low frequencies at the opposite end. The figure shows the approximate locations of various frequencies.

The right side is a schematic diagram of the response of hair cells along the basilar membrane, as a group of overlapping signals.

The process is similar to an analog radio receiver, which receives the frequency signal to which it is tuned from a nearby radio station. The more the receiver and station frequencies do not match, the more unstable and distorted the signal will be, regardless of its strength. There are upper (and lower) levels of the frequency range beyond which hair cells cannot receive signals and we cannot hear anything.

Sample rate and audible frequency spectrum

I’m sure you’ve heard many times that frequencies 20 Hz to 20 kHz are the audible range of the human ear. It is very important to understand how scientists obtained such numbers.

First, we measure the “hearing threshold” across the entire audio range for a group of listeners. This allows us to construct a curve that represents the quietest sound the human ear can hear at any given frequency, measured under ideal conditions in healthy ears. An anechoic environment, accurate calibration of breeding equipment, and rigorous statistical analysis are an easy part of the experiment. Auditory concentration is lost very quickly, so the test must be performed while the subject is not tired. As a result, there are many breaks and pauses, and testing can take from several hours to several days, depending on the methodology.

What is the DSD music format?

What is the DSD music format?

DSD

What is the DSD music recording format?

DSD

What is the PCM audio recording format?
Where to download or buy music recordings in DSD format
The DSD format is becoming increasingly popular as Internet connections accelerate and home appliances become more and more powerful. And if 10 years ago the mp3 format dominated unconditionally in computers and players, today there are more and more lovers of music recorded with high quality. Recently, several dozen people have asked me where to download music in DSD format. This really surprised me, because no one had asked me those questions before. It’s good that people are starting to value quality over compactness of audio files.

What is the DSD music recording format? What are DSD files?
To clearly imagine what DSD files are, we need to know a slightly different format: PCM.

I apologize for the fact that the video lesson is in English, unfortunately in Russian there are no good videos explaining the method of recording data in digital format in PCM format. However, even with a basic level of knowledge of the English language, you will understand almost everything, because the video has illustrative drawings.

What is the PCM audio recording format?
PCM is a pulse code modulation used to digitally represent analog signals. It is the standard form of digital audio in computers, CDs, digital telephony, and other digital audio applications. PCM is an uncompressed audio format that stores lossless audio data and often serves as the input for other types of audio files. For example, CD music is stored in PCM format with a resolution of 16 bit / 44.1 kHz.

The DSD format was developed by Sony and Philips in 1999 and was intended to replace the standard CD format. At that time it was called Super Audio CD or SACD. Today this format is called DSD.

You may have come across the name of the DXD format from time to time, and no, this is not a typo. This is a different type of file. DXD stands for Digital eXtreme Definition and is used when the original DSD signal has been converted to 24-bit / 352 kHz PCM. We can say that DXD is very high resolution PCM.

But going back to DSD, how good is it? The What Hi-Fi site finally put an end to this question, saying:

“There is no exact method or way to compare PCM and DSD formats one by one due to different encoding algorithms, but if you need a general idea then DSD is roughly equal in recording resolution to PCM 24 bit / 88.2 kHz. . As for the dynamic range, for DSD it is about 120 dB. In comparison, the dynamic range for a CD recording cannot exceed 96 dB, while a 24-bit / 192 kHz recording can theoretically have a dynamic range of up to 144 dB. ”

Where can I download or buy music recordings in DSD format?
We have talked very briefly about how DSD music recordings differ from others. In short, quality. Today, DSD recordings are of the highest quality and if your equipment supports DSD music playback, I recommend that you listen to only those records. Even with mid-range headphones, you’ll notice a difference, not to mention high-end headphones.

Audio pcm what. Digital sound: DSD vs PCM Part 3

Audio pcm what. Digital sound: DSD vs PCM Part 3

What is DSD Audio? [Sound Quality, DSD vs PCM]

Retrieve a “digit” analog signal

But digitizing an analog signal is half the battle. To listen to digital music, you must reverse convert. First, let’s see how to convert a digital DSD broadcast to sound. As we already know, this stream is a high frequency bi-level signal (2.8 MHz or more), the average value of this signal changes with the audio frequency. That is, if the approach to solving the problem is as simple as possible, you need to filter out all the high-frequency components of the DSD stream, leaving only a useful sound signal (frequencies up to 20 … 22 kHz). This is done using an analog low pass filter (LPF). The simplest LPF is an RC chain.

As you can see, the resulting graph only vaguely looks like the original sinusoid. But let’s not forget that we “applied” the simplest filter, improving the filter circuit can achieve an almost total absence of high frequency noise and obtain an analog sound with good quality indicators.

To restore an analog signal from a digital PCM, just an analog low-pass filter is not enough, you must first decrypt the digital data, for this, digital-to-analog converters (DACs) are used. They are of different types, but it is beyond the scope of this article to describe them all. Let’s dwell on the 2 most common types of sound technology. First of all, this is the so called ladder type DAC (also called multibit). As you probably guessed, such a DAC converts a PCM digital data stream into a stream of audio signal values ​​that look like a ladder on the graph (Figure 6). As with DSD, it is imperative to use an analog filter to smooth out the jogging.

Often these converters use intermediate oversampling of the digital PCM signal at higher frequencies (eg 192 kHz): this reduces the “steps”, allowing for simplification of the analog filter circuit.

The second type of DAC, delta-sigma, uses oversampling at even higher frequency values ​​with a simultaneous reduction of the bit depth to one bit. Doesn’t it look like anything? This is a familiar DSD signal! We have already discussed how to further process such a signal and convert it to analog.

PCM and DSD application, advantages / disadvantages
Where can we find each of the encoding methods? PCM format is very common: CDDA discs, DVD audio, MP3 files, FLAC, ALAC, AAC, sound in movies, and so on, it is easier to say when it is not PCM. Super Audio CD, DSD, DSF, DFF files are in DSD format. What is better? What format will we get a better sound from?
The articles dedicated to the DSD format describe many advantages over PCM, but are all the advantages described true, or are they myths invented for laymen who do not understand the technical component to recover the market densely occupied by the PCM format? Let’s briefly review the list.

conclusions
So should you choose DSD or PCM? There is no single answer and it cannot be: PCM 24 bit 92 kHz and DSD128, for example, are very similar in quality characteristics, and these characteristics are better than the equipment on which these formats will be played, which means a further increase in the quality of digital formats for playback at this stage is not practical. When evaluating the quality of sound in different high definition formats, subjective sensations come to the fore, because the human brain is not eaten by the same quality: the design of the equipment, its cost and, most importantly, the well-being and the The listener’s moods have a much greater effect on the sensation of listening to music. Therefore, choose what you like personally and do not impose your opinion on others. Happy listening everyone!
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Audio pcm what. Digital sound: DSD vs PCM Part 2

Audio pcm what. Digital sound: DSD vs PCM Part 2

DSD Vs PCM - Real Competitors? | Headfonics Audio Reviews

First, let’s answer the question, what is digital sound? How is it different from analog? In short, in mathematical terms, an analog audio signal is a continuous function, a digital audio signal is a discrete function. What does that mean?

Analog signal
If we draw in our imagination a graph of a sinusoid (this is how a sound wave is most often represented): then, no matter how we magnify it, trying to see all the details, we will always see a smooth and uniform line – this is an analog audio signal

Analog audio (recording) has many parameters that can be used to evaluate its quality. Consider the three most important: frequency range, dynamic range, distortion.

Digital sound. How many myths revolve around this phrase. How many disputes have arisen between lovers of comfort and digital quality and supporters of “live airy” vinyl sound multiplied by “warm tube” sound. In addition, there is a lot of controversy among lovers of “numbers”: is 16×44.1 enough or is 24×192 necessary? Which is better: multibit or delta sigma? CDDA or SACD? PCM or DSD? In this article, I will try to explain the basics of digital sound in simple language and will also expand in more detail on comparing two types of encoding of an analog to digital signal: DSD and PCM.

First, let’s answer the question, what is digital sound? How is it different from analog? In short, in mathematical terms, an analog audio signal is a continuous function, a digital audio signal is a discrete function. What does that mean?

Analog signal
If we draw in our imagination a graph of a sinusoid (this is how a sound wave is most often represented): then, no matter how we magnify it, trying to see all the details, we will always see a smooth and uniform line – this is an analog audio signal

Analog audio (recording) has many parameters that can be used to evaluate its quality. Consider the three most important: frequency range, dynamic range, distortion.

The frequency range is a set of frequencies contained in a sound. It is generally accepted that the frequency range of human hearing is 20 … 20,000 Hz (sometimes 16 to 22,000 Hz is indicated). The frequency range of the music itself is of no interest in terms of quality assessment (for example, the frequency range of the same plane taking off will be very wide and the tenor’s vocal part will be much narrower). A qualitative parameter, say, of an earphone is the potential frequency range, and it is estimated using the amplitude frequency characteristic (AFC). The ideal frequency response, a straight line across the entire range of hearing frequencies, means that the sound source does not amplify or attenuate any individual frequencies, meaning that the extracted sound matches the original.

Dynamic range (DD) is the difference between the quietest and loudest sound. Loudness is measured in decibels (dB). It is generally accepted that the maximum volume that does not cause injury to a person is 130 dB, the sound of an airplane taking off, and the minimum audible volume, 5 … 10 dB, is at the level of the rustling of the leaves in low wind conditions. Naturally, it will be impossible to distinguish the rustle of leaves against the background of a plane taking off, and listening to music at a level of 130 dB is extremely unpleasant. Therefore, it is generally accepted that a comfortable DD for listening to music is 80 … 100 dB.

The distortion is nothing more than a deviation of the signal from the original.

Principles of digital sound presentation
What happens when I digitize analog audio? We will not delve into the technical aspects, we will analyze everything, as they say, on paper: for this we will draw our imaginary “ideal” sinusoid and measure the value of the signal at regular intervals (this process is called sampling or quantization): we will obtain a certain sequential set of values ​​- this will be our digital signal obtained by the pulse code modulation (PCM) method

The two main parameters of PCM signal quality are frequency and bit depth. Frequency is the number of measurements per second, the more, the more accurate the signal is transmitted. Frequency is measured in Hertz: 44100 Hz, 192000 Hz, etc. Bit depth: the number of possible values ​​of the signal value (precision of the value transmission). The more options, the more accurate the signal will be. Bit depth is measured in bits: 16 bits (65,536 possible values, DD 96 dB), 24 bits (16,777,216 values, DD 144 dB), etc.

Audio pcm what. Digital sound: DSD vs PCM Part 1

Audio pcm what. Digital sound: DSD vs PCM Part 1

DSD vs. PCM

What is PCM

DSD & PCM

Let’s start with the fact that PCM (Pulse Code Modulation) is initially older, the first mentions of its successful use date back to the middle of the last century and are associated, like many technological advances, with the defense industry, that is, with the Navy radars. As for home use, first of all, it is a well-known CD with a sampling frequency of 44.1 kHz and a 16-bit quantization level.

What is DSD
DSD (Pulse Density Modulation) is a format developed by Sony and Philips at the end of the last century and intended for the digital archiving of analog phonograms. The physical medium of this format is SACD. In fact, there is only one similarity between these two formats, both are digital, which for the user means the possibility of making unlimited copies without loss. As for the difference, relative to the field of graphic design, it is roughly the same as raster and vector graphics. And if it is even more artistic, like cross stitch and watercolor. In both cases, an image is obtained, but the method of its creation and, as a result of perception, are completely different.

What is the difference?
PCM, even because of its age, is much more studied, it has much better compatibility with a large number of very different devices, it implies the possibility of editing (equalization, division into frequency bands, transformations). DSD is actually a closed format, you can record to it, you can play it, that’s it. However, it is inherently much closer to the original analog signal.

Which is better?
The first and most important conclusion is that from a technical point of view, the formats are far apart in terms of implementation methods, but they are often practically indistinguishable in practical use, that is, in the sound of the final file. We are talking only about minor differences in the nuances of the musical presentation. So, all things being equal, when choosing the next file to download and play, it’s best to focus on the source material. If you are looking to digitize an analog then DSD will probably be preferable and will retain more nuances from the original. If this is a remastering of a digital recording previously made in PCM, then it would make more sense for it to stay in this domain.

Digital sound. How many myths revolve around this phrase. How many disputes have arisen between lovers of comfort and digital quality and supporters of “live air” vinyl sound multiplied by “warm tube” sound. In addition, there is a lot of controversy among lovers of “numbers”: is 16×44.1 enough or is 24×192 necessary? Which is better: multibit or delta sigma? CDDA or SACD? PCM or DSD?

First, let’s answer the question, what is digital sound? How is it different from analog? In short, in mathematical terms, an analog audio signal is a continuous function, a digital audio signal is a discrete function. What does that mean?

Digital audio formats or how sound is stored on a computer

Digital audio formats or how sound is stored on a computer

Digital Audio Formats

Today there are about three dozen common digital audio formats. Why you need to create so many types of sound files to store one type of content and how to manage all this, you will learn from this material.

Audio format developments | Digital audio | How to Create Digital Media  Infographics Using ConceptDraw PRO | Audio Infographic

Surely many users prefer to use their home computer not only as a workhorse, but also as a multimedia center, where they can watch movies or family photos, as well as listen to their favorite music. Although compact digital players or mobile phones are certainly more suitable for listening to musical compositions, but unlike them, a computer can not only play music.

No matter how big the built-in memory of your music player is, it will most likely be difficult to store your entire music library on it. Plus, you can create, edit, organize, and search for music with your PC. Also, don’t forget that there are around three dozen common digital audio formats today, and most players are far from omnivorous and can only play a few of them.

So why do you need to create so many music formats to store one type of content? The fact is that, in the vast majority of cases, the sound is stored in “compressed” form, since one minute of uncompressed composition occupies about 10 MB on the hard disk. On the one hand, this seems not to be much, but on the other, if you are a music lover and your collection consists of several hundred or even thousands of songs, then it is clear that the sound must be compressed to reduce the space it occupies in electronic media.

Various special algorithms are used to compress music files, which subsequently determine the structure and presentation of the audio data, or so-called digital audio file formats. All audio formats can be divided into three groups: uncompressed audio formats, lossless compression, and lossy compression.

No compression
One of the most widespread formats related to this type is the well-known WAV. The sound of files with this extension is stored without compression or changes. It is true that much more space is required to store uncompressed files and therefore WAV is more widely used only in professional audio and video applications, where the sound should not have a loss of quality before processing. Keeping ordinary musical compositions in this form is unwarranted waste.

To play WAV files, you do not need any special software, as all media players understand this format, including the standard Windows Media audio player built into the Windows system.

Another format used to store uncompressed audio that is worth mentioning is Apple’s development called AIFF (Audio Interchange File Format). As you may have guessed, it is most commonly used on Macintosh computers running Mac OS X.

Lossless compression (lossless)
Lossless compression algorithms for audio files work on the principle of conventional file cabinets. They do not provide the highest level of compression (40 to 60%), while they have virtually no effect on sound quality. It is also worth noting that in this case, the encrypted data can be fully restored to its original form. Therefore, the use of lossless compression is most often used when it is important to keep the compressed data identical to the original.

The most popular audio formats in this group are FLAC (Free Lossless Audio Codec), APE (Monkey’s Audio), WMA (Windows Media Lossless), and ALAC (Apple Lossless Audio Codec). Each has its own pros and cons. For example, the APE codec offers slightly better compression gains, while FLAC is more common. In general, all true music lovers store their music collections in lossless formats, as they do not remove any data from the audio stream and the files created with these codecs can be listened to even on high-quality stereos.

To play lossless compressed formats, as a rule, third-party players (except WMA) are used, such as MPlayer, foobar, AIMP, Winamp, VLC and others, since all the necessary codecs are already built into them. Another option is to separately install an additional codec pack (for example, K-Lite), after which you can listen to files in lossless format from almost any audio player.

Lossy compression
This is the most popular group of algorithms that provides the maximum audio compression ratio (up to 10 times or more). However, unlike previous formats, the audio file loses quality here, and how much depends

Mp3 vs OGG digital audio format (OGG VORBIS)

OGG digital audio format (OGG VORBIS)

MP3 vs Ogg Vorbis Compression - Visual Comparison - YouTube

This digital audio format, like MP3, also uses lossy compression.

The OGG format is based on a psychoacoustic model, but the mathematical processing is fundamentally different from MP3.

The OGG format has the main advantage: with the same bit rate, this format offers superior quality!
It is the OGG digital audio format that is popular and is second only to MP3 in prevalence. This format is compatible with all computer programs for playing sound files and all operating systems.

But unlike MP3, it has less support at the hardware level.
Not all stereos, audio players, and DVD players support this format.

Considering the advantages of the OGG digital audio format, it is promising. And it’s the future in the digital audio format market. The development of recent years shows that the OGG digital format is becoming more and more popular.

OGG digital audio format (OGG VORBIS)
By the way, in the comments to the article on the MP3 digital audio format, most wrote that as a better alternative, you need to use this compressed digital format.

The advantages of this digital format:
1. More modern: it was presented to the public in 2002.

2. This format is the most MODERN among the competitors of the MP3 format.

3. It has a series of innovations, such as:

– Advanced and better quality psychoacoustic model and mathematical compression model;
– High quality sound reproduction with the same bit rate as MP3;
– Supports multi-channel audio;
– can handle up to 255 different channels.
– variable bit rate encoding is possible, which makes the compression algorithm more flexible;
– a wider range of bit rates and sample frequencies available for compression (2 to 192 kHz);
– the ability to adjust audio compression more flexibly, achieving the necessary balance, just for you, between recording quality and file size;
– has the most flexible tag management system among all competitors – textual explanations for the audio file, including: the name of the song, the album it is included in, the author, the year of release and much more useful information (it has practically no length limit and can be in different languages).
The MP3 implementation only allows 2 channels (stereo sound) to work.
Considering the fact that modern sound devices have long crossed the threshold of the two channels (for example, the 5.1 format, which uses five speakers and a subwoofer), this difference is very important for consumers and listeners , especially in modern movies and computer games.

Comparison of common audio formats: which one to use?

Comparison of common audio formats: which one to use?

WMA

Audio files come in all types and sizes. And although we are all familiar with MP3, how about AAC, FLAC, OGG or WMA? Why are there so many standards? Which ones should you care about and which ones can you ignore?

WMA

It’s actually pretty simple once you understand that all audio formats fall into three main categories. Once you know which category you want, all you have to do is choose the format in that category that best suits your needs.

Uncompressed audio formats
Uncompressed audio is exactly what it sounds like: real sound waves that have been captured and digitized without any additional processing. As a result, uncompressed audio files tend to be the most accurate, but they take up A LOT of disk space, around 34MB per minute for 96kHz 24-bit stereo.

PCM
PCM stands for Pulse-Code Modulation, the digital representation of raw analog audio signals. Analog sounds exist as signals, and to convert a signal into digital bits, the sound must be sampled and recorded at specific intervals (or pulses).

Therefore, this digital audio format has a “sample rate” (how often a sample is taken) and a “bit depth” (how many bits are used to represent each sample). There is no compression. Digital recording is a nearly accurate representation of analog audio.

PCM is the most common audio format used on CDs and DVDs. There is a subtype of PCM called linear pulse code modulation, where samples are taken at linear intervals. LPCM is the most common form of PCM, so at this stage the two terms are almost interchangeable.

Wav
WAV stands for Waveform Audio File Format (also called Audio for Windows at one point, but not anymore). It is a standard developed by Microsoft and IBM in 1991.

Many people assume that all WAV files are uncompressed audio files, but this is not entirely true. WAV is actually a Windows container for audio formats. This means that a WAV file can contain compressed audio, but it is rarely used for this.

Most WAV files contain uncompressed PCM audio. The WAV file is just a wrapper for PCM encoding, which makes it more suitable for use on Windows systems. However, Mac systems can generally open WAV files without any problem.

AIFF
AIFF stands for Audio Interchange File Format. Like Microsoft and IBM developed WAV for Windows, AIFF is a format that Apple developed for Mac systems in 1988.

Like WAV files, AIFF files can contain various types of audio. For example, there is a compressed version called AIFF-C and another version called Apple Loops that use GarageBand and Logic Audio, and they all use the same AIFF extension.

Most AIFF files contain uncompressed PCM audio. The AIFF file is simply a wrapper for PCM encoding, which makes it more suitable for use on Mac systems. However, Windows systems can generally open AIFF files without any problem.

audio file signal format

Lossy compressed audio formats
Lossy compression is a form of compression that loses data during the compression process. In the context of audio, this means that we sacrifice quality and fidelity for file size. The good news is that most of the time you won’t be able to tell the difference.

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However, if the audio is compressed too much or too often, you will start to hear artifacts and other oddities that become more and more noticeable.

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MP3
MP3 stands for MPEG-1 Audio Layer 3. It was released in 1993 and quickly gained popularity, eventually becoming the world’s most popular audio format for music files. There’s a reason we have “MP3 players” and not “OGG players” …

The main pursuit of MP3 is to remove all audio data that exists outside the audible range of most normal people and reduce the quality of sounds that are not easy to hear, and then compress all other audio data from the as efficiently as possible.

Almost every digital audio capable device in the world can read and play MP3 files, whether we are talking about PC, Mac, Android, iPhone, Smart TV or anything else. When you need versatility, MP3 will never let you down.

Please note that MP3 is not the same as MP4

although their similar names may indicate otherwise.

ACC
AAC stands for Advanced Audio Coding. It was developed in 1997 as the successor to MP3.