Advantages and disadvantages of digital sound


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Advantages and disadvantages of digital sound

digital sound

From the point of view of a normal user, there are many benefits: the compactness of modern storage media allows you, for example, to transfer all the disks and records in your collection to a digital representation and save for many years in three small ones.

Digital Sound

one-inch hard drive or on a dozen or two CDs; you can use special software and thoroughly “clean” old records from reels and records, removing noise and crackle from their sound; It can also not only correct the sound, but also beautify it, add richness, volume, restore frequencies. In addition to the listed manipulations with sound at home, the Internet also comes to the rescue of the audio lover. For example, the network allows people to share music, listen to hundreds of thousands of different Internet radio stations, and also to show your sound creativity to the public, and for this you only need a computer and the Internet. And finally, recently, a large number of various portable digital audio equipment has appeared, the capabilities of which even for the most average representative often make it easy to carry a collection of music with a duration equivalent to tens of hours on the road. . .

From a professional’s point of view, digital audio offers truly endless possibilities. If the previous radio and sound studios were located on several tens of square meters, now they can be replaced by a good computer, which surpasses ten of those studios combined in capabilities and is much cheaper than one in terms of cost. This removes many financial barriers and makes recording more accessible to both the professional and the simple amateur. Modern software lets you do what you want with sound. Previously, various sound effects were achieved with the help of ingenious devices that did not always live up to technical thinking or were simply handcrafted devices. Today, the most complex and hitherto unimaginable effects are achieved by pressing a couple of buttons. Of course,

Of course, digital technology has its drawbacks, too. Many (professionals and amateurs) note that the analog sound was heard with greater intensity. And this is not just a tribute to the past. As we said before, the digitization process introduces a certain error in the sound, in addition, various digital amplifier equipment introduces the so-called “transistor noise” and other specific distortions. Perhaps there is no precise definition of the term “transistor noise”, but we can say that they are chaotic oscillations in the high frequency region. Although the human hearing aid can perceive frequencies up to 20 kHz, it appears that the human brain picks up higher frequencies. And it is on a subconscious level that a person still feels analog sound cleaner than digital.

However, the digital representation of data has an indisputable and very important advantage: with a saved medium, the data it contains does not distort over time. If the magnetic tape becomes degaussed over time and the recording quality is lost, if the record is scratched and pops and crackles are added to the sound, then the CD / hard disk / electronic memory is readable (if preserved) or not , and there is no aging effect. It is important to note that we are not talking about audio CDs here (CD-DA is a standard that establishes the parameters and format for recording on audio CDs), since even though it is a carrier of digital information, the effect of aging still won’t get away. This is due to the peculiarities of storing and reading audio data from an audio CD.


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Digital Audio Storage Methods – PART 2

Digital Audio Storage Methods – PART 2

Digital Audio

Due to the use of the new SBR (Spectral Band Replication) technology, the codec performs notably better than other formats at low bit rates; however, the quality of encoding at medium and high bit rates is generally inferior to the quality of almost all the codecs described. Therefore, MP3 Pro is more suitable for streaming audio over the Internet, as well as creating previews of songs and music. however, the quality of encoding at medium and high bit rates is often lower than the quality of almost all the codecs described.

Digital Audio

Therefore, MP3 Pro is more suitable for streaming audio over the Internet, as well as creating previews of songs and music. however, the quality of encoding at medium and high bit rates is often lower than the quality of almost all the codecs described. Therefore, MP3 Pro is more suitable for streaming audio over the Internet, as well as creating previews of songs and music.

Speaking of the methods of storing sound in digital form, one cannot help but remember the data carriers. The familiar audio CD, which appeared in the early 1980s, has become mainstream in recent years (which is associated with a sharp reduction in the cost of media and drives). And before that, digital data carriers were magnetic tape cassettes, but not ordinary ones, but specially designed for so-called DAT recorders. Nothing extraordinary: tape recorders are like tape recorders, but the price for them has always been high, and that pleasure was not too difficult for everyone. These recorders were used primarily in recording studios. The advantage of such recorders is that despite the use of familiar media, the data on them was stored in digital form and there was practically no loss during reading / writing on them (which is very important for studio processing and recording. sound storage). Today, a large number of different storage media have appeared, in addition to the usual compact discs. The media are improved and every year they become more accessible and compact. This opens up great opportunities in the field of creating mobile audio players. Today a large number of different models of portable digital players are already on sale. And we can assume that this is far from the peak of the development of this type of technology. This opens up great opportunities in the field of creating mobile audio players. Today a large number of different models of portable digital players are already on sale. And we can assume that this is far from the peak of the development of this type of technology. This opens up great opportunities in the field of creating mobile audio players. Today a large number of different models of portable digital players are already on sale. And we can assume that this is far from the peak of the development of this type of technology.

Digital audio storage methods

Digital audio storage methods

digital audio

There are many different ways to store digital audio. As we said, digitized sound is a set of signal amplitude values ​​taken at regular intervals. Thus, first, a block of digitized audio information can be written to a file “as is”, that is, a sequence of numbers (amplitude values). In this case, there are two ways to store information.

DIGITAL AUDIO

The first is PCM (Pulse Code Modulation), a method of digitally encoding a signal by recording the absolute values ​​of the amplitudes (there are signed or unsigned representations). In this way, the data is recorded on all audio CDs.

The second method – ADPCM (Adaptive Delta PCM – adaptive relative pulse code modulation) – records signal values ​​not at all, but in relative changes in amplitudes (increments). Second, you can compress or simplify the data so that it takes up less memory than when it was written “as is.” There are also two ways here.

Lossless Data Encoding (Lossless Encoding) – is an audio encoding method that enables data recovery from a fully compressed stream. This method of data compaction is used when it is essential to maintain the quality of the original data. For example, after mixing sound in a recording studio, the data should be saved to the file in its original quality for possible later use. Today’s lossless encoding algorithms (for example, Monkeys Audio) can reduce the volume of data occupied by 20-50%, but at the same time ensure one hundred percent recovery of the original data from the data obtained after compression. Such encoders are a kind of data archivers (such as ZIP, RAR and others), only designed for audio compression.

There is also a second encoding path, which we will dwell on in a little more detail, lossy data encoding (lossy encoding). The purpose of such encoding is to achieve the sound similarity of the reconstructed signal to the original by any means with the least possible amount of packed data. This is achieved through the use of various algorithms that “simplify” the original signal (eliminating “unnecessary” details for the hearing impaired), leading to the fact that the decoded signal is no longer identical to the original, but only sounds similar. There are many compression methods, as well as programs that implement these methods. The most famous are MPEG-1 Layer I, II, III (the latter is the well-known MP3), MPEG-2 AAC (advanced audio encoding), Ogg Vorbis, Windows Media Audio (WMA), TwinVQ (VQF), MPEGPlus, TAC and others. On average, the compression ratio provided by such encoders is in the range of 10-14 (times). It should be noted that at the heart of all lossy encoders is the use of the so-called psychoacoustic model, which is simply involved in “simplifying” the original signal. More precisely, the mechanism of such encoders analyzes the coded signal, in the process of which the signal sections are determined, in certain frequency regions of which there are nuances inaudible to the human ear (masked or inaudible frequencies), after which are removed. of the original signal. Therefore, the degree of compression of the original signal depends on the degree of its “simplification”; Strong compression is achieved by “aggressive simplification” (when the encoder “considers” various nuances unnecessary), such compression naturally leads to strong quality degradation, as not only imperceptible but also significant sound details can be removed .

As we said, there are a lot of modern lossy encoders. The most common format is MPEG-1 Layer III (known as MP3). The format gained its popularity quite deservedly: it was the first widespread codec of its kind, achieving such a high level of compression with excellent sound quality. Today, there are many alternatives to this codec, the choice is up to the user. Unfortunately, the scope of the article does not allow us to provide tests and comparisons of existing codecs here, however, the authors of the article will allow themselves to provide some information that is useful when choosing a codec.

So the advantages of MP3 are the widespread use and a fairly high encoding quality, which is objectively improved thanks to the development of various MP3 encoders by enthusiasts (for example, the Lame encoder). A powerful alternative to MP3 is the Microsoft Windows Media Audio codec (.WMA and .ASF files).

MP3, FLAC, WAV, ALAC: the differences between audio formats

Digital audio formats

Digital Audio

Today, most people listen to music completely digitally. The differences between digital audio formats like WAV, FLAC, MP3, and ALAC are not clear to everyone. We put the facts together.

Digital audio formats

While vinyl is booming and CD sales are slowly but surely falling, today’s music is often heard without any physical medium. Whether you use your smartphone or digital audio player, you can move forward with digital audio formats on the go. After all, no one today wants to carry a Discman and multiple CDs with them when they typically have a powerful pocket computer in the form of a smartphone that can play digital music files. But what are the differences between the individual file formats and what are their advantages and disadvantages?

WAV and AIFF: the uncompressed ones

The Wave container format (.wav) was developed by Microsoft. Saves uncompressed audio content, so files require a lot of storage space (2 minutes can take 20MB of space. WAV is especially important when recording and editing audio content. The downside of .wav files is that they don’t metadata is required (about, Title Artist) can be stored,
the equivalent developed by Apple AIFF (.aif) Due to the fact that Apple computers are very common in music production, this audio format is very common there.

MP3, AAC, WMA, Ogg-Vorbis – compressed to save space, but not lossless

The MP3 file format (.mp3, named for the MPEG-1 Audio Layer 3 compression codec) developed by the Fraunhofer Institute in the 1980s is probably the best-known digital audio format. It gave the MP3 player its name, and for a long time music was digitized almost exclusively as MP3, for example, on the extremely popular and now illegal file-sharing networks around the turn of the millennium. The advantage of MP3 is the small amount of storage space required: on average, it takes up one-tenth the size of the original file. However, one disadvantage that should not be neglected is that it is lossy – frequencies that are inaudible to humans are removed to drastically reduce the memory required. To what extent this affects the sound, you can compare Flac with MP3 Read.

AAC (Advanced Audio Coding) is a successor to the MP3 format, offering slightly better sound quality. Apple continues to mainly offer songs in this audio format on the iTunes store.

WMA stands for Windows Media Audio (.wma), as the name suggests, a development by Microsoft. .Wma is also a lossy compression file format.

A somewhat rarer audio format is Ogg-Vorbis (.ogg), where Vorbis is the music compression technology and .ogg is the container format. Like MP3, .ogg is also lossy, but requires less storage space and better quality.

FLAC / ALAC / WMA lossless – the lossless

Lossless formats were developed to preserve all sound information while keeping the amount of memory required small. With all file formats, the required memory is reduced to about half the original file. With audio conversion software, the file can be converted to other lossless formats, something unthinkable with lossy formats. This is why lossless file formats are popular for archiving music collections in a space-saving way.

FLAC – Free Lossless Audio Code (.flac) is a free audio format, so it is not owned by any major corporation. ALAC: Apple Lossless Audio Codec (.alac) is Apple’s lossless file format, while Microsoft also has its own development on the market with WMA Lossless.

Basics of digital audio

Basics of digital audio:

Before the computer can record, manipulate, and reproduce sound, sound must be transformed from an audible analog form to a computer-acceptable digital form, using a process called analog-to-digital conversion (ADC). Once the sound data has been stored as bytes in the computer, the power of the computer’s CPU can be used to transform this sound in thousands of ways. Finally, when you are ready to listen to the result, the digital-to-analog conversion (DAC) process transforms the sound bytes back into an analog electrical signal from the speakers.

Sampling: Analog to Digital Conversion

Given an analog signal, discrete values ​​of its amplitude are taken at small time intervals, obviously the more reliable the reproduction the more samples per second are taken. These obtained values ​​are assigned a digital value that the computer can understand and process as required. We can use 8 or 16 bit words, thus obtaining 256 or 65536 different combinations and obtaining higher resolution.

 

SAMPLE FREQUENCY: According to the Nyquist theorem, it is possible to accurately repeat a waveform if the sampling frequency is at least twice the frequency of the component with the highest frequency. The highest frequency that the human ear can perceive is close to 20 kHz, so the 44.1 kHz sampling rate of sound cards is more than enough. This value is the one used today by CD audio players.

SAMPLE SIZE: The sample size controls the dynamic range that can be recorded. For example, 8-bit samples limit the dynamic range to 256 steps (50 dB range). In contrast, a 16-bit sample has a dynamic range of 65,536 steps (90 dB range) a substantial improvement. The human ear perceives a whole world of differences between these two sample sizes. Ears are more sensitive to detecting differences in pitch than intensity, but are even more sensitive to the strength of sound.

From the previous processes we can get an audio file, such as (and since it is the best known), a WAV audio file. It is the own format of Windows. They can be 8 or 16 bit with sampling rates of 11,025 kHz, 22.05 kHz, or 44.1 kHz and generally have good sound quality.

Digital audio compression

It could be assumed that all you have to do to get good sound is to record at the 44.1 kHz speed limit with 16-bit (2-byte) samples. The only problem that appears if recording in stereo, sampling simultaneously on the left and right channels at 44.1 kHz, a one minute sound sample needs a 10.58MB storage space. This involves using large disk spaces to store these sound files. Many compressed file formats (codecs) have been developed that enable high-quality recording without the need for so much disk space.

Most common audio formats:

With the simple objective of listing a series of codecs used by different operating systems to perform audio compression. Later, a more complete description of the most used is made: MP3.

Therefore, some of the most used are:

Advanced Audio Coding (AAC): used by Apple computers. More efficient than MP3.

Audio for Unix (AU): Acoustic standard for the JAVA programming language.

Windows Media Audio (WMA)

Ogg Vorbis: It is free, open and not patented.

Atrac: compression and playback technology for minidisc.

 

The codec par excellence: the MP3

Its origin and current

The abbreviations MP3 respond to the abbreviation of MPEG (Moving Picture Expert Group) 1 Layer 3, which is a perceptual coding algorithm. This among others was developed by the Moving Picture Expert Group (MPEG) (http://www.cselt.it/mpeg/) together with the Fraunhofer Institute of Technology (http://www.ipa.fhg.de/english/ ).

Moving Picture Expert Group is an ISO / IEC research committee. MPEG is in charge of the international development of compression, decompression, processing and encoded rendering standards for movies, audio and the combination of both. It is a non-profit institution created in 1988, which brings together 300 experts from 20 countries three times a year.

Introduction to digital audio

Introduction to digital audio

Digital audio is the representation of sound signals through a set
of binary data. A complete digital audio system usually begins
with a transceiver (microphone) that converts the pressure wave that represents the
Sound to an analog electrical signal.
This analog signal goes through an analog signal processing system, in
which can be made limitations on frequency, equalization, amplification and
Other processes such as compassion. Equalization aims
counteract the particular frequency response of the transceiver used of
so that the analog signal closely resembles the original audio signal.


After analog processing, the signal is sampled, quantified and encoded. The
sampling takes a discrete number of analog signal values ​​per second
(sampling rate) and quantification assigns discrete analog values ​​to those
samples, which means a loss of information (the signal is no longer the same
than the original). The encoding assigns a sequence of bits to each value
discrete analog The length of the bit sequence is a function of the number of
analog levels used in quantification. The sampling rate and the
number of bits per sample are two of the fundamental parameters to choose from
when you want to digitally process a certain audio signal.
Digital audio formats try to represent that set of samples
digital (or a modification) of them efficiently, so that it is optimized
depending on the application, either the volume of the data to be stored or the
processing capacity necessary to obtain the starting samples. In
in this sense there is a very extended audio format that is not considered audio
digital: the MIDI format. MIDI does not start with digital sound samples, but
stores the musical description of the sound, being a representation of the
score of them.
The digital audio system usually ends the reverse process to that described. From
the stored digital representation is obtained the set of samples that
represent. These samples go through a process of digital analog conversion
providing an analog signal that after processing (filtering,
amplification, equalization, etc.) affect the output transceiver (speaker)
which converts the electrical signal to a pressure wave that represents the sound.

Fundamental parameters of digital audio

The basic parameters to describe the sequence of samples it represents
The sound are:
ƒ The number of channels: 1 for mono, 2 for stereo, 4 for sound
quadraphonic, etc.
ƒ Sampling rate: The number of samples taken per second in each
channel.
ƒ Number of bits per sample: Usually 8 or 16 bits.
As a general rule, multichannel audio samples are usually organized in
frames A plot is a sequence of as many samples as channels,
each one corresponding to a channel. In this sense the number of samples per
second matches the number of frames per second. In stereo, the channel
Left is usually the first.