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discussion on dynamic range compression.

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discussion on dynamic range compression.

Dynamic range compression is a scapegoat for poor musical sound, but heavily compressed music isn’t a new trend – listen to Motown albums from the sixties. The same can be said of the Led Zeppelin classics or the younger Wilco and Radiohead albums.

Records, especially older ones that were recorded and produced before 1982, were less likely to get mixed up and get louder. They reproduce natural music with a natural dynamic range that is preserved on record and lost in most standard or high definition digital formats.

Of course, there are exceptions – listen to Steven Wilson’s recently released album from MA Recordings or Reference Recordings and you’ll hear how good digital sound can be. But this is rare, most modern recordings are tall and compressed.

Music compression has been the subject of serious criticism lately, but I would say that almost all of your favorite recordings are compressed. Some of them are less, some more, but they are still compressed. Dynamic range compression is a scapegoat for poor musical sound, but heavily compressed music isn’t a new trend – listen to Motown albums from the sixties. The same can be said of the Led Zeppelin classics or the younger Wilco and Radiohead albums. Dynamic range compression reduces the natural ratio between the loudest and lowest recorded sounds, so whispers can be as loud as screams. It’s pretty hard to find pop music from the last 50 years that hasn’t been compressed.

I recently had a nice chat with Tape Op founder and editor Larry Crane about the good, bad and bad aspects of compression. Larry Crane has worked with bands and artists such as Stefan Marcus, Cat Power, Sleater-Kinney, Jenny Lewis, M. Ward, The Go-Betweens, Jason Little, Eliot Smith, Quasi, and Richmond Fontaine. He also runs the Jackpot recording studio! in Portland, Oregon, home to The Breeders, The Decemberists, Eddie Vedder, Pavement, REM, She & Him and many, many more.

Crane agreed with my arguments, but added: “The compression conversation needs to be approached from two different sides: are we talking about compressing the entire track in the mixing and mastering process, or compressing individual music tracks (instruments and vocals) in the recording and mixing process? “That’s right, compression is applied at all stages of music production, so some of the dynamic range may have been lost long ago when the mastering engineer performed the last run. If you don’t have access to the multitrack master copy, the two tracks after mixing, and the final master copy, then you won’t be able to understand why the recording sounds like this.

As an example of surprisingly unnatural sound, but still great songs, I cite Spoon They Want My Soul’s album, released in 2014. Crane laughs and says he listens to it in the car because he sounds great there. Which brings us to another answer to the question why music is compressed: because compression and the extra “clarity” make it sound better in noisy places.

Larry Crane at work. Photo by Jason Quigley

When people say they like the sound of an audio recording, I think they like music, as if sound and music are inseparable terms. But for me, I differentiate these concepts. From a music lover’s point of view, the sound may be harsh and raw, but that won’t matter to most listeners.

Many are in a hurry to accuse mastering engineers of abusing compression, but compression is applied directly during recording, during mixing, and only then during mastering. If you have not been personally present at each of these stages, then you will not be able to know what the instruments and voices sounded like at the beginning of the process.

Crane was on fire: “If a musician deliberately wants to make the sound crazy and distorted like Guided by Voices records, then there is nothing wrong with that: desire always outweighs sound quality.” The performer’s voice is almost always compressed, the same goes for bass, drums, guitars, and synthesizers. Compression keeps vocal volume at the desired level throughout the song or stands out slightly from other sounds.

Compression done correctly can make the drums sound more lively or intentionally strange. In order for the music to sound good, you must be able to use the instruments necessary for this. That’s why it takes years to figure out how to use compression and not go overboard.

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Compression of audio signals

Compression of audio signals.

Audio data compression

Audio compression is widely used in professional and consumer digital audio products, such as compact disc (CD), digital audio type (DAT), mini disc (MD), digital compact cassette (digital compact cassette – DCC), versatile disc digital (DVD), digital audio broadcasting (DAB) and MP3 audio products from M <Picture Experts Group – (MPEG).

Audio Data Compression

While voice compression in telephony, in particular cellular telephony, necessary to save bandwidth and save time and battery, has led to the development of many voice compression standards, personal algorithms are applicable to voice signals and consumer of a wider frequency band. Voice and audio compression schemes can be conveniently classified according to applications, reflecting some measure of acceptable quality.

Adaptive Differential PCM (ADICM).

Using the past data to measure (i.e., quantify) the new ones, we went from conventional pulse code modulation (PCM) to differential (differential PCM – DPCM). In DPCM, a prediction of the next sampled value is generated based on the previous values. The quantizers are called instantaneous quantizers or non-memory quantizers because the digital transformations are based on a single (current) input sample. These properties were unequal source levels and dependent sample values. The correlation characteristics of a source can be represented in the time domain by sampling its autocorrelation function and in the frequency domain by its power spectrum. If we study the power spectrum Gx (f) of a short-term speech signal, as shown in Figure 9.2, then we see that the spectrum has a global maximum in the vicinity of 300 to 800 Hz and decays at a rate 6 to 12 dB / octave. This operation is performed on the legend and the comparison contour, the upper contour of the encoder is shown in Figure 13.2. The encoder adjusts its predictions by adding the predicted value and the prediction error.

This model, which uses a 12-lead speech synthesizer, has found application in children’s conversation games.

Compression algorithm

MPEG

The International Organization for Standardization (ISO) and the Motion Picture Experts Group (MPEG) have developed the audio compression standard for synchronized video signals known as MPEG. This scheme combines the properties of MUS1CAM (Masking Pattern Adaptive Universal Subband Integrated Coding and Multiplexing) and ASPEC (Adaptive Spectral Perceptual Entropy Coding). The scheme uses three levels (codes) of increasing complexity and improvement of subjective performance. The input sampling frequencies are 32, 44.1 and 48 kHz, and the bits are output at 32 to 192 kbps (monaural) or 64 to 384 kbps (stereo). The standard supports single channel mode, stereo mode, dual channel mode (for bilingual audio programs), and optional collaborative stereo mode. In the latter mode, the two encoders for the left and right channels can support each other using common statistics to reduce the bit rate of the audio signal even more than is possible with mono transmission.

Level III of the MPEG / ISO (MP3) standard achieves a higher frequency resolution that is very close to critical human resolution.

Digital audio encoding

Digital audio encoding

Digital audio encoding

In fact, one or another digital form of representation of analog audio signals is already a coding method – a sequence of numbers that describes an analog audio signal is itself a digital code.

Digital Audio Encoding

However, the encoding that we are going to talk about now is something else. Now let’s look at the methods of encoding digital audio signals.

A digitized audio signal “in its pure form” is a fairly accurate, but not the most compact, way of recording the original analog signal.

Judge for yourself. To obtain complete information about the original analog signal in the frequency range 0-20 kHz (in the audible frequency range), the analog signal must be sampled at a frequency of at least 40 kHz. Therefore, the CD – DA standard (the standard for recording data on audio CDs familiar to all) establishes the following encoding parameters: recording of two or one channel in PCM format with a sampling frequency of 44.1 kHz and a 16-bit quantization bit depth. One hour of music in this format takes up approximately 600 MB of space (60 minutes * 60 seconds * 2 channels * 44100 samples per second * 2 bytes per sample = approximately 605 MB). Taking into account that, for example, the music collection of an ordinary music lover may have 5,000 tracks with an average length of about 3 minutes each, the amount of memory required to store it in its original digital form is quite significant. Awesome. Therefore, storing relatively large amounts of audio data, ensuring fairly good sound quality, requires the use of various “tricks” to compress the data.

In general, all existing methods for encoding audio information can be conditionally divided into only two types.

1. Lossless data compression (“Lossless Encoding”) is a method of encoding (compacting) digital audio information, which enables one hundred percent recovery of the original data from the compressed transmission (the term ” original data “here means the original form of the digitized audio data). This method of data compression is used in cases where one hundred percent absolute preservation of the quality of the original audio data is required. Lossless compression algorithms that exist today can reduce the volume of data occupied by 20-50% and at the same time guarantee a 100% recovery of the original digital material from the compressed data. The operating mechanisms of such encoders are similar to the operating mechanisms of general data archivers, such as ZIP or RAR, but at the same time they are specially adapted to compress audio data …. Lossless encoding While it is ideal in terms of preserving the quality of audio materials, it cannot provide a high level of compression.

2. There is another more modern way to compact data. This so-called lossy data compression (Engl. “Lossy encoding”) The purpose of encoding is to achieve the highest data compression rate by all means while keeping sound quality at an acceptable level. The idea behind lossy encoding is based on two simple underlying considerations:

original digital audio data is redundant: it contains a lot of unnecessary information that is useless to the ear, which can be removed, thereby increasing the compression ratio;
Requirements for the sound quality of audio material may vary and depend on specific purposes and areas of use.
Lossy encoding is therefore called “lossy”, which results in the loss of some of the audio information. Such encoding leads to the fact that the decoded signal, when reproduced, sounds similar to the original, but in reality it is no longer identical to it. Most lossy coding methods rely on the use of the psychoacoustic properties of the human auditory system, as well as various tricks associated with resampling and resampling the signal. In frequency, during the compression process, the encoder analyzes the audio data to identify various details of the sound that can be ignored. Disguised frequencies, inaudible and inaudible sound details can be sacrificed for a higher compression ratio. Where intelligibility is only important in sound (for example, in telephony, where the presence of frequencies above 4 kHz is not necessary), the audio information during the encoding process undergoes a serious “simplification”, which, together with the use of successful “smart” quantifiers and “greedy” data compression algorithms.

Audio compression algorithms for streaming purposes.

Audio compression algorithms for streaming purposes.

The problem of transmitting the necessary number of audio channels through a network of limited capacity forces us to resort to audio compression. Despite the use of modern digital technologies, compression negatively affects sound quality and causes additional delay in signal transmission.

Currently, there are two fundamentally different approaches to compressing audio signals. This article will provide a general comparison between these two different compression principles. Also presented are graphs of the frequency response (amplitude frequency characteristic) of the sound sample in its original uncompressed form and after one cycle of encoding and decoding using MPEG Layer II and Enhanced apt-X.

Algorithms like MPEG and AAC use encoding using a psychoacoustic model of sound perception. Another approach is time encoding using Adaptive Differential PCM (ADPCM) in algorithms like Enhanced apt-X.

Linear PCM audio
Before compression, the audio is generally digitized in linear PCM format at 32 kHz, 44.1 or 48 kHz with a resolution of 16 or 24 bits.

The analog signal will be digitized in uncompressed digital PCM. The digital inputs of the codecs use oversampling to ensure conversion without timing issues. The uncompressed PCM signal is our benchmark for comparing compressed audio files.

MPEG Layer ll compression
MPEG 1 Layer ll is a widely used format. This is a typical example of a psychoacoustic perception coding algorithm that analyzes the incoming signal and compares it to a theoretical model to determine what frequency and what time domain information could be lost. The need to analyze the audio signal results in a mandatory delay, typically greater than 30 ms.

In theory, high compression ratios can be achieved, but even with relatively low compression, MPEG can seriously degrade audio quality. In Fig. 2 shows the frequency response after one pass of MPEG encoding of the source file.

Be aware of frequencies that are lost or distorted from the original PCM audio.

Compression Enhanced Apt-X
Enhanced apt-X uses ADPCM audio processing technology. The signal is divided into four frequency bands that can be processed at a quarter of the original sample rate using a variable bit rate and a variable quantization step. Since all processing is based on a time domain method, there is no delay other than the actual processing time required.

As a result, a 4: 1 compression ratio retains the entire frequency content of the original signal with a coding delay of less than 3 ms. Frequency response graph in Fig. 3 shows the result of one pass encoding / decoding using Enhanced apt-X at 256 kbps and illustrates the high fidelity of Enhanced apt-X compared to the original uncompressed signal.

How Enhanced apt-X Works
The improved apt-X encoding algorithm passes the original PCM data through a specially designed two-stage Q-mirror filter to divide the signal into four subbands and reduce the clock to 1/4 of the original clock frequency. The quantization procedure consists of processing four sub-signals to reduce each signal from 16 bits to 7 bits in subband 1, 4 bits in subband 2, 3 bits in subband 3 and 2 in subband 4.

The inverse quantizer and prediction scheme uses the above values to predict the size of the next signal. This value is compared to the actual signal and the “difference” is measured. The encoder transmits this measured “difference” signal to the decoder. Each subband is processed in parallel and the output of the string quantizer and predictor is encoded with a predetermined resolution. The processing output of the four subbands is multiplexed into a single 16- or 24-bit enhanced apt-X signal. Then additional data and sync data are added to it for streaming.

Comparison by main points
MPEG / AAC encoding is destructive: frequencies are lost during the encoding process.
Enhanced apt-X encoding is non-destructive, as every frequency present in the original signal is stored in the encoded and decoded signal.
MPEG and AAC suffer from the concatenation effect: repeated encoding and decoding cycles rapidly degrade audio quality.
Enhanced apt-X is resistant to concatenation – repeated encode and decode cycles do not cause any noticeable degradation in sound quality.

Digital audio formats

Digital audio formats

Digital Audio

The digital audio format is a format for presenting audio data used in digital audio recording, as well as for additional storage of recorded material on a computer and other electronic media, so-called audio media.

digital audio

The audio file (a file containing a sound recording) is a computer file consisting of information about the amplitude and frequency of sound, saved for later playback on a computer or player.

Varieties of digital audio formats.

There are several concepts of audio format.

The digital representation of the audio data depends on how the digital-to-analog converter (DAC) quantizes. In sound engineering, two types of quantization are currently the most common:

pulse code modulation

sigma delta modulation

Quantization bit depth and sample rate are often specified for various audio recording and playback devices as a digital audio rendering format (24-bit / 192 kHz; 16-bit / 48 kHz).

The file format determines the structure and presentation characteristics of the audio data when stored on a PC storage device. To eliminate the redundancy of the audio data, audio codecs are used, with the help of which the audio data is compressed. There are three groups of audio file formats:

uncompressed audio formats like WAV, AIFF

lossless compressed audio formats (APE, FLAC)

lossy compressed audio formats (mp3, ogg)

Modular music file formats are highlighted. Created synthetically or from prerecorded live instrument samples, they are primarily used to create modern electronic music (MOD). Also, this can be attributed to the MIDI format, which is not a sound recording, but at the same time, using a sequencer, it allows you to record and play music using a certain set of commands in the form of text.

Digital audio media formats are used for both mass distribution of sound recordings (CD, SACD) and professional sound recording (DAT, minidisc).

For surround sound systems, sound formats can also be distinguished, which are mainly multichannel sound accompaniments for movies. These systems have complete format families from two major competitors, Digital Theater Systems Inc. – DTS and Dolby Laboratories Inc. – Dolby Digital.

The format is also called the number of channels in multichannel sound systems (5.1; 7.1). This system was originally developed for movie theaters, but has since been expanded for home theater systems.

Mobile Hi-Fi: Understanding Music Formats

Sony celebra 40 años del Walkman con lanzamiento de dispositivo retro

In recent years, the topic of high-quality sound “on the go” is more relevant than ever. Digital players are making a comeback, DAC technologies that were previously only available to expensive audiophile systems are now slipping into your pocket, and streaming services are beginning to stream in high-resolution quality. It’s time to find out if you need headphones for the price of a car and what really affects sound quality.

iPod

History
The first truly mobile player was the Sony Walkman in 1979. Then it revolutionized music. The mere fact that the music could be played out of his pocket seemed fantastic. Cassette recordings were loud, the tape could be chewed, and to save energy it might be fun to rewind with a pencil. However, the Walkman was an innovative product that was very commercially successful for more than 20 years.

Literally a few years later, in 1983, the Sony Discman appeared, which played CDs, an ultra-modern format at the time that has not lost its relevance to this day. The sound was much better than the cassette, but with each movement, the music froze, as the laser head could not stay on the optical track. However, the main problem with portable CD players was size: these players turned out to be much larger than cassette players. Sony’s next format to solve the size problem, MiniDisc, failed miserably.

The first MP3 players appeared in the late 1990s. They played heavily compressed MP3 files that could barely fit on the 32MB internal memory. Apple’s iPod revolutionized portable digital music, introduced by Steve Jobs as “1000 songs in your pocket.” Later, the first truly popular store where you could buy and download music appeared: the iTunes Store.

1st generation Apple iPod.
The next stage in the development of portable music was smartphones and streaming services in the late 2000s. And this is “Over 30 million songs in your pocket.” No more buying compressed music for the price of a CD: For a small monthly fee, you get almost all the music on the planet.

Now
Sound quality has become the trend of the current decade. People are tired of low quality music. Vinyl record sales (!) Are breaking all records. In 2016, more than 3.2 million records were sold in the UK, an increase of 53% over 2015, while digital music purchases fell 30%.

High-resolution audio formats emerged from the professional recording environment, which began to appear on streaming services. Smartphones have learned to work with external DACs that are capable of delivering Hi-End sound straight out of your pocket. In portable headphones, the technology came from internal monitors that musicians use during live concerts. And finally, the players have made a comeback, but not from cheap plastic, but from metal, glass, and high-quality audio processors inside. Not a bad time to be a music fan!

Portable Hi-Fi system
To get good sound, simply plugging an expensive headset into your smartphone is not enough. . Of course, the sound will be better compared to the “plugs” that come with the phone, but to get really good sound you will need to go through the entire chain, from the recording studio to your ear.

Digital recording formats

Digital recording formats

Digital Recording

Television video equipment using digital tape recording methods has been supplied for several years.

Digital Recording

However, the majority of television workers in the world still do not have a sufficiently clear idea of what a “digital” is, why it is needed and if it is worth working on it. The media specializing in television technology have focused on image quality, comparing the quality of digital formats, digital compression, artifacts, etc. Communication with television engineers shows that negative opinion often comes from the experience of working in incorrect settings, faulty equipment, or due to ignorance of the features and subtleties of the latest technology. That is why those who are now dedicated to television continue to be confused with the new concepts and prefer to use the old Betacam SP or the inferior, but familiar S-VHS. And some television executives, wanting to keep up and trying to make sense of the new teams, get misconceptions due to lack of information and focus their attention on insignificant details, missing highly profitable opportunities. Meanwhile, digital formats are developing rapidly, the range of relevant equipment is expanding,

So, in the process of developing video recording formats, the following main features have been improved:
– picture quality;
– operational capabilities;
– recording density and cost of 1 minute of recording;
– weight and size parameters of the videotape;
– the cost of purchasing and operating the equipment.

Image quality
Image quality generally refers to resolution, that is, the number of vertical lines reproduced. Of course, this is a cursory assessment, as there are many other, no less important parameters that are as perceptible to the human eye as the readability of the line. This review will also take into account the signal-to-noise ratio of the path, the quality of the composite, Y / C and component encoders / decoders, the effectiveness of the fight against tape drops. To a large extent, the evaluation of this parameter was quite subjective.

Operational capabilities
This concept includes everything related to the operation of the device in the system, operational functions, ease of use, integration capabilities, presence of interfaces and inputs and outputs, certain configuration characteristics, etc.

Recording density, cost per minute of recording, and video tape size and weight parameters
These are important parameters for the video recording format. They take into account three factors: the size and weight of the videotape, the duration of the recording, and the unit cost of one minute of recording. The higher the capacity of the cassette and / or the smaller its size, and / or the lower the unit cost, the higher the estimate.

Expenses
This parameter takes into account the cost of equipment, maintenance and spare parts. According to our system, a high score corresponds to a lower total cost of ownership and maintenance of equipment of one format or another.

In each new video format, developers strive to improve these indicators, but improvement in one indicator often occurs at the expense of others. But it must be admitted that the total level of indicators for all categories is growing from format to format.
Let us consider the video formats that are used in television centers in our country. Format features are rated on a 10-point scale, with ratings in parentheses after each feature. The effectiveness of the format will be determined by the total score.

Analog formats

Format Q (Ampex, NZTM)
Home television switched to video recording, using Q-format Kadr-3PM video recorders, using 2 “wide tape videotape as the carrier. Naturally, there was no Q-format camcorder at all. : mobile TV stations and cameras Format features:

1. The Q format provides high image quality (6 MHz bandwidth). Signal / noise ratio 40 dB. A complete composite video signal is recorded on tape in the SECAM standard, so no transcoding is required for broadcast playback. (Score 8)
2. The opportunities are minimal. VCRs can only be used by qualified engineers; preparing for playback requires at least five minutes of careful adjustment of the CAP modes and parameters of each of the four video heads. There is no search for an image with preview, markup editing is done in motion only.

Formats: what is digital sound Part 3

Formats: what is digital sound Part 3

digital sound

JointStereo, which is one of the features of MP3, means that instead of encoding stereo as two independent channels, it encodes calls. center channel and the difference from the original stereo channels.

DIGITAL SOUND

Many audio components on stereo channels are the same, and encoding them on the common channel allows you to free up additional bandwidth for more detailed encoding of the difference, leading to improved quality.

Be sure to mention the variable bit rate or VBR. This means that the encoder changes the compression ratio on the fly, depending on the nature of the sound. This approach results in a smaller total file size or, if quality requirements increase, the same file size may produce better sound.

MP3 Pro

Introduced in 2001, the MP3 Pro codec was developed by Coding Technologies in association with Thomson Multimedia. It is MP3-based and as a result turned out to be fully MP3 backward compatible and only partially forward compatible. It uses SBR (Spectral Band Replication) technology, so the codec provides good quality at low bit rates. However, the encoding quality at medium to high bit rates is inferior to almost all other codecs. As a result, MP3 Pro is used more for streaming on the Internet and demonstrating snippets of new musical compositions.

WMA

The WMA codec, or Microsoft Windows Media Audio, is a serious alternative to MP3. Files in this format have the extensions .WMA and .ASF, have a clear advantage over MP3 at low data rates (bitrates) and lose it when the data feed rate to the codec is increased.

Based on WMA, the WMA DRM standard has been developed to provide copy protection so appreciated by record companies. Files based on this format can be recorded on playback devices such as MP3 flash players, but cannot be copied from there.

ATTRAC

ATRAC (Adaptive TRansform Acoustic Coding) is based on a stereo audio format with 16-bit quantization and a sample rate of 44.1 kHz. When compressed, each frame is divided into 52 frequency bands, resulting in a transmission rate of 292 kbps (1: 5 compression). Used in MiniDisk system.

Ogg Vorbis

The Ogg Vorbis format is a relatively new universal lossy audio recording format. It belongs to the same type of audio compression formats as MP3 and WMA, and the psychoacoustic model that describes the characteristics of the human ear, according to which compression is performed, is similar in principle to MP3. The radical difference of this format was the mathematical processing and the practical implementation of this model. In this format, the maximum threshold sample rate is not 44 kHz as in MP3, but 48, which theoretically improves the sound quality. It should also be noted that the theoretical number of channels is not limited to two, as usual, but reaches 255. Files encoded in this format are smaller than the same MP3 files. The spread of the format was slowed by insufficient support from hardware manufacturers.

Formats: what is digital sound Part 2

Formats: what is digital sound Part 2

Digital Sound

Historically, digital sound was undoubtedly the initiative of company engineers who adopted Philips-developed Audio-CDs, also called CDDA – Compact Disk Digital Audio.

DIGITAL SOUND

As a result, digital audio media led to a massive transition in recording studios to digital DAT tape recorders and digital editing equipment with S / PDIF and other interfaces. And then digital sound began to penetrate deeper into our lives from CD players, and as it was transmitted via S / PDIF, it became digital switches, equalizers, and noise reduction systems. Today this series ends with Dolby Digital surround sound processors.

Who needs it

CDDA’s sound quality is satisfactory for most end users, ie listeners, but the amount of data required to present sound in this way is critical. As a result, several compressed digital audio formats appear, one of which is the old MS ADPCM, and among which are quite acceptable Sony ATRAC, PASC or Fraunhoffer MP3. Each of the encoding methods has an important characteristic – the bit rate, with which the compressed information enters the decoder when the audio signal is restored.

For example, when you talk on a cell phone, the sound of your voices is digitally converted and compressed, degrading its performance. Various algorithms compress speech hundreds of times, preserving the basic characteristics.

Let’s move on to specific audio file formats and audio compression formats. The most common format today is, of course, MP3. However, historically, to understand the evolution of sound formats, it is necessary to start with a different type of file, with the extension .WAV.

Variety of formats

Wav

It is the primary format for many, many digital audio playback systems and is used as a standard audio file format on personal computers. In addition, it has a strong set of specifications, which has grown considerably lately. Its full name is Microsoft RIFF / WAVE – Resource Interchange File Format / Wave – Resource Interchange File Format / Waveform, and it was created by Microsoft and Intel engineers. In turn, WAV is short for Waveform Audio File Format.

Apple AIFF

This type of file is standard for Apple Macintosh systems and sound processing systems based on it. Apple AIFF stands for Audio Interchange File Format: an audio interchange file format, it is somewhat similar to WAV. Its peculiarity is that it allows additional information to be placed next to the sound wave, in particular WaveTable samples (examples of the instrument sound together with synthesizer parameters), which improves the quality of the final result. Although today Apple computers are capable of playing files in almost any format, including MP3.

RAW

Yes, this is not just the image format in which some digital cameras take pictures. In fact, RAW is the call. “Pure Digitization”, which does not contain a title and only contains a sequence of samples of a sound wave. Typically, the scan is stored in 16-bit format.

MP3

The most popular compression format today is MP3. The MP3 (MPEG Layer 3) format was developed, after several intermediate formats, by the Fraunhofer Institute in Germany. Actually, the .MP3 format is based on fooling the human ear. After some research, it turned out that human hearing tends to adapt to the appearance of new sounds, which is expressed in an increase in the hearing threshold. Therefore, some sounds are capable of masking (that is, making them subjectively inaudible) others. So in this format, some of the sounds that, according to the corresponding theory, are made inaudible, are simply removed from the general sound. The resulting “semi-finished product” is then encoded using the Hoffman method. Be sure to note that in MP3 format, programs that compress the sound of the original are not standardized, that is, each competent programmer can implement their own compression scheme. And only decoders meet the standards, which leads to the fact that the quality of MP3 playback does not always depend on the player that plays this file. Due to the different skills and predilections of implementers of various encoders, some of them are better at handling symphonic music, some with rock and metal, some with rap and rave, etc.

Formats: what is digital sound

Formats: what is digital sound

Digital Sound

Sound plays an increasingly important role in the modern world, having long since separated itself from the close link with the image that emerged during the heyday of television and cinema.

digital sound

Modern multimedia equipment has the widest possibilities not only for playback, but even for changing the sound. It is no longer a dead record, a static reproduction of events from the past, firmly imprinted on its medium. The most important role in transforming our ideas about sound was played by the development of a digital method of recording sound, turning it into a data stream that can be easily and naturally operated with modern devices.

The basics of “numbers”

Each of the multimedia devices on sale today, be it a CD player, a voice recorder or a flash memory player, uses many different types of presentation of data streams, which are then converted into sound. And even more sound formats used for professional purposes have been invented. An inexperienced buyer is forced to gather information about designations on boxes and devices from a variety of sources, often receiving incorrect information or even more confusion.

Almost all devices in the “Portable Audio” section of the ZOOM.CNews.ru catalog support multiple sound formats at the same time, and many devices that do not belong in this category are also tagged with support for playing sound files. To help our reader, we decided to create a short glossary of abbreviations and talk about the most common formats. We plan to leave it open for updates and modifications, adding new formats and describing in more detail the advantages and disadvantages of the already common or forgotten ones.

A little theory

To begin with, remember that digital sound is nothing more than a collection of numbers. The determining factor is the system by which sound as air pressure is converted into data streams and encoded for further processing and reproduction. Consequently, digital sound is usually included in computer files with various extensions, which more often (but not always) can determine their format. And the same concept of format can have, paradoxically, two meanings. First, the format may exist as a general characteristic that includes both the type and the physical characteristics of the medium (disk or cassette), the method of recording, the principles of encoding, and protection against errors. Second, the format can only be understood as the method of audio encoding and compression, as standard means are used for transfer, for example,

Analog sound, unlike digital, is reproduced on analog devices and has a number of significant differences. While not a data stream, analog audio is represented as a continuous electrical signal that represents a change in the sound wave. To translate it into digital format, the sound is “digitized”, that is, it is divided into certain segments, in which the numerical value of the amplitude is fixed at that moment. We will not delve into the principles of digital sound creation, but it is absolutely necessary to note that the more often a sound segment is divided and its characteristics described, the clearer and more complete the sound image itself is created.

This process generates a large amount of data that describes the sound, and it is clear that each digital audio format is nothing more than a compromise between the need to present the sound as loudly as possible and the memory limitations of a computer or device. Of reproduction.

A little more theory. In most cases, the human ear perceives sound with a frequency no higher than 22,000 Hz and, to describe it fully in digital form, a sampling frequency of at least 44.1 kHz is required. Since it is absolutely impossible to determine the value of the signal at any given moment, during digitization quantization occurs, that is, the replacement of the actual values of the signal by approximate values. The more levels of audio quantization, the more accurately the signal level is described. As a result, each standard CD carries an audio signal with a sampling frequency of the same 44.1 kHz and a 16-bit quantization level, and in some devices, sampling is done at 48 kHz.

The benefits of digital audio

And now, briefly on why this algorithm was developed. Digital sound has enormous advantages over analog, although we must not forget its certain disadvantages. The main value of digital sound is the possibility of infinitely long storage and playback