Audio and Video Compression Basics

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Audio and Video Compression Basics

As we rely more and more on digital media, understanding the basics of audio and video compression becomes increasingly important. Compression is the process of reducing the size of digital files without sacrificing too much quality. Without compression, media files would take up a lot more space on our hard drives, making it difficult to store and share them. In this article, we’ll explore the fundamentals of audio and video compression and how it works.

Understanding Audio Compression

Audio compression is the process of reducing the dynamic range of an audio signal. Dynamic range is the difference between the quietest and loudest parts of a sound recording. Compression reduces this difference, making the quieter parts louder and the louder parts quieter. This is useful for improving the overall balance of a mix, and also for preventing distortion when the loudest parts of a recording exceed the maximum level of the recording medium.

Compression can be applied during recording or in post-production, using software tools like mp4gain. When done properly, compression can improve the clarity and punch of a recording, making it sound more polished and professional. However, overuse of compression can lead to a loss of detail and a “squashed” sound that lacks dynamics.

As musician David Byrne said in his book “How Music Works”:

“A good mix is one where the listener can hear and feel everything that the musicians and the engineer intended to be there.”

Understanding Video Compression

Video compression is the process of reducing the size of a video file by removing redundant or unnecessary data. This is done by encoding the video using a codec, which stands for “coder-decoder”. Codecs use complex algorithms to analyze each frame of a video and compress it in a way that minimizes the loss of quality.

There are two types of video compression: lossless and lossy. Lossless compression reduces the size of a video file without any loss of quality, but it’s not as effective as lossy compression in terms of file size reduction. Lossy compression, on the other hand, sacrifices some quality to achieve a smaller file size. The level of quality loss depends on the amount of compression applied.

When it comes to video compression, there are many factors to consider, including the resolution, bit rate, and frame rate. By adjusting these parameters, you can find the right balance between file size and quality for your particular needs.

As filmmaker and author Robert Rodriguez once said:

“Filmmaking is a chance to live many lifetimes.”

Compression Techniques for Audio and Video

There are many compression techniques used in audio and video, each with its own strengths and weaknesses. In audio, the most common type of compression is called “peak compression”, which reduces the volume of loud sounds that exceed a certain threshold. Another type of compression, called “multi-band compression”, divides the audio signal into multiple frequency bands and applies compression to each band separately.

For video compression, the most popular codecs are H.264 and HEVC (High-Efficiency Video Coding). H.264 is widely used for streaming video on the internet, while HEVC is more efficient but requires more processing

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Audio Compression Techniques: Understanding the Basics

What is Audio Compression?

Audio compression is the process of reducing the size of digital audio files by removing redundant or unnecessary information, while maintaining the perceived quality of the original sound. This is done by using various algorithms that analyze and modify the audio data in a way that reduces its file size.

Types of Audio Compression Techniques

There are two main types of audio compression techniques: lossy and lossless.

Lossy Compression

Lossy compression algorithms are used to achieve high compression rates, but at the cost of some loss in quality. In lossy compression, some of the original audio data is discarded or modified in a way that reduces its size. The amount of data that is removed or modified depends on the compression algorithm used.

Some popular lossy compression algorithms include MP3, AAC, and WMA. These algorithms are commonly used for music streaming, online radio, and other applications where high compression rates are necessary.

Lossless Compression

Lossless compression algorithms are used to compress digital audio files without losing any information. These algorithms are designed to reduce the size of the file by removing redundancies in the data, but without modifying any of the original information.

Some popular lossless compression algorithms include FLAC, ALAC, and WAV. These algorithms are commonly used for high-quality music streaming and for archiving music collections.

How Audio Compression Works

Audio compression works by analyzing the original audio data and then modifying it in a way that reduces its size while maintaining its quality. This is done using various mathematical algorithms that compress the data.

The most common way to compress audio data is to use perceptual coding. This method takes advantage of the human ear’s limitations in hearing certain frequencies and sounds. By removing these sounds, the audio data can be compressed without the listener noticing any loss in quality.

Another method of audio compression is predictive coding. This method uses mathematical algorithms to predict the next sample in a waveform based on previous samples. The difference between the predicted sample and the actual sample is then compressed and stored.

Why Audio Compression is Important

Audio compression is important because it allows us to store and transmit audio data more efficiently. This means that we can store more audio files on our devices and transmit audio data faster over the internet. Without audio compression, it would be impossible to stream music or podcasts over the internet.

12 Common Questions About Audio Compression Techniques

1. What is the difference between lossy and lossless audio compression?

Lossy compression algorithms are designed to achieve high compression rates at the cost of some loss in quality, while lossless compression algorithms are designed to compress audio files without losing any information.

2. Which audio compression algorithm should I use?

The choice of audio compression algorithm depends on the intended use of the audio file. Lossy compression algorithms like MP3 and AAC are commonly used for music streaming and online radio, while lossless compression algorithms like FLAC and ALAC are commonly used for high-quality music streaming and archiving.

3. How much does audio compression affect the quality of the original sound?

The amount of quality loss in audio compression depends on the compression algorithm used and the degree of compression applied. Lossy compression algorithms generally result in some loss in quality, while lossless compression algorithms do not.

4. How can I tell if an audio file has been compressed?

You can usually tell if an audio file has been compressed by looking at its file extension. Lossy compressed files usually have extensions like MP3, AAC

Methods of compression and compression of audio signals Part 3

Methods of compression and compression of audio signals Part 3

Audio Compression

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 relies 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 the 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 the decoders obey 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 abilities and predilections of implementers of various encoders, some of them are better at handling symphonic music, some at rock and metal, some at rap and rave, etc.

JointStereo, which is one of the features of MP3, means that instead of encoding stereo as two independent channels, it encodes the call. center channel and the difference from the original stereo channels. Many stereo channel audio components 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 reduction in the final file size or, if quality requirements increase, the same file size produces 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 it 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.

The MPEG-4 audio standard does not require a single or small set of highly efficient compression schemes, but rather a complex set to perform a wide range of operations, from low-quality speech coding to high-quality music and audio synthesis.

The MPEG-4 family of audio coding algorithms ranges from low quality voice (up to 2 kbps) to high quality audio (64 kbps per channel and higher).

RAW – Yes, it is not just the image format in which some digital cameras write photographs. In fact, RAW is the so-called. “Pure digitization”, which does not contain a title and contains only a sequence of samples of a sound wave. Typically, the scan is stored in 16-bit format.

Shorten is one of the first lossless codecs to appear. For a long time the project “slept sweetly.” However, in 2007, it began to develop again.

TTA (True Audio) – Finally about the most interesting. TTA is being developed by a team of our compatriots. And, I must say, the result of their work is impressive. All in order.

The codec is still quite young, but despite this it contains all the necessary features. We won’t list them again, we’ll just note that the format only lacks support for streaming audio over the network.

The format is open, as well as the source codes of the encoder program. There are compiled versions for Mac and Linux. There should be no compatibility issues during playback either, because there are already plugins for all popular players, as well as DirectShow filters for Windows Media Player. There is a plugin for Adobe Audition, which is important for musicians. For the past 4 years, hardware support has even appeared on players!

WAV – This is the primary audio format for many, many digital audio playback systems and is used as a standard audio file format on personal computers.

Compression and compression methods for audio signals Part 2

Compression and compression methods for audio signals Part 2

audio compression

FLAC is a member of the Xiph.Org codec family. By the way, it also includes the well-known ogg vorbis, one of the best lossy music compression algorithms. As a container for audio data, of course, OGG (files with the extension .ogg) and another open source container – Matroska (files with the extension .mka) are used.

It should be noted right away that both the FLAC format and algorithm are fully open. They are not patented, so they can be used completely free of charge in any program. This is the reason for the wide support for FLAC in players – any serious gamer has a plugin for FLAC. In addition, there are hardware mp3 players that support the FLAC codec.

The FLAC encoder is compiled for most platforms in use, so there should be no compatibility issues on alternative Windows operating systems.

FLAC supports tags in its own “FlacTags” format. There is the ability to encode multi-channel audio, a great advantage over Monkey’s Audio. The format supports any sample rate in the range of 1 Hz (!) To 65,535 Hz. Audio bit depth from 4 (!) To 32 bits.

FLAC is believed to be the most efficient use of system resources when decoding (playing) audio compared to other lossless codecs. Unfortunately, this is achieved at the expense of a significant increase in encoding (compression) time.

The FLAC website is regularly updated and new versions of the codec are released. Overall, FLAC is without a doubt the leader in terms of development activity. This may make it the main format in the future. Well, let’s see …

FLAC is the best option for storing high quality music.

MIDI (Musical Instrument Digital Interface) is a standard for hardware and software that allows you to play (and record) music by executing / recording special commands, as well as the format of the files that contain those commands. The playback device or program is called a MIDI synthesizer (sequencer) and is actually an automatic musical instrument.

Unlike other formats, it does not store the digitized sound, but sets of commands (played notes, links to played instruments, variable sound parameter values) that can be played differently depending on the playback device. The convenience of the MIDI format as a data representation format enables devices that produce automatic arrangements according to given chords, as well as 3D sound visualization applications. Additionally, these files tend to be orders of magnitude smaller than digitized audio of comparable quality.

Monkey’s Audio is a popular lossless digital audio encoding format. Distributed for free along with open source and a suite of encoding and playback software, as well as plugins for popular players. Monkey’s audio files use the following extensions: .ape to store audio and .apl to store metadata. Despite being open source, Monkey’s Audio is not free, as its license imposes significant restrictions on its use.

Audio files compressed with the Monkey audio codec have the extension ‘APE’; As you can see, the monkeys are present not only in the logo or the name (from English monkey: monkey, primate).

The average bit rate in an audio file is 600 to 700 kbps; compare with 128 kbps in MP3. Average compression is 40-50%, depending on the genre of music: if classical or jazz pieces are compressed in the best way, then compositions in the style of trash-metal or something similar “electronic noise” will show the worst result. . For codecs with acceptable quality loss, compression is approximately 80%.

There are four levels of compression. Maximum compression may seem like the only correct solution, although the compression time is quite long. However, you must also take into account the resource consumption of the system that plays the file; for the most compressed file, it is relatively high.

The .APE format provides tag support for searching for songs in your music collection. Another advantage is the verification of the integrity of the file during decoding. Recovery of original compressed .APE wav files is supported.

Monkey’s Audio has a graphical interface for Windows, in other words, a convenient window program to manage the encoding process. The rest of the codecs require the use of the command line or third-party interfaces.

Compression and compression methods of audio signals

Compression and compression methods of audio signals (types, differences, use)

Audio Compression

Basics of the analog-to-digital conversion principle, sound conversion and compression method, existing sound storage formats. Programs to convert and process sound and audio files. Application of these programs in linguistic research.

Bit rate is the amount of information per unit of time. In general, the bit rate is the number of bits that we spend encoding a sound with a duration of 1 second.

Analog-to-digital converter (ADC): A device that converts an input analog signal into a binary code (digital signal). The reverse conversion is done using a DAC (digital-to-analog converter, DAC). Typically, an ADC is an electronic device that converts voltage into a binary digital code. However, some non-electronic devices with digital output must also be classified as ADCs, such as some types of angle-to-code converters. The simplest one-bit binary ADC is a comparator.

The circuit to convert an audio signal from analog to digital:

Sampling is the transformation of continuous images and sound into a set of discrete values in the form of codes.

Quantization is the process of aligning a set of musical notes to a grid.

Compression (compression) of audio data is a process of lowering the bit rate by reducing the statistical and psychoacoustic redundancy of a digital audio signal.

The underlying idea behind all lossy audio compression techniques is to neglect the subtle details of the original sound that are beyond the reach of the human ear.

Codec (CoDec) is an abbreviation for compressor and decompressor. Basically, a codec is a collection of files, drivers, and libraries required to package a video or audio file into a compressed format and play the compressed file.

Formats:

AAC (Advanced Audio Coding) is an audio file format with less quality loss when encoding than MP3 of the same size. The format also allows you to compress without losing the quality of the source (ALAC AAC profile).

AAC (Advanced Audio Coding) was originally created as a successor to MP3 with improved encoding quality. The AAC format, officially known as ISO / IEC 13818-7, was released in 1997 as the new seventh part of the MPEG-2 family. There is also the AAC format known as MPEG-4

Apple AIFF: This file type is standard for Apple Macintosh systems and sound processing systems built on top of 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 you to place additional information along with 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 of almost any format, including MP3.

FLAC (Free Lossless Audio Codec) is a popular free codec for audio compression. Unlike lossy Ogg Vorbis, MP3 and AAC codecs, it does not remove any information from the audio stream and is suitable for both daily listening and archiving of audio collection. Today, the FLAC format is compatible with many audio applications.

Digital audio compression methods

Digital audio compression methods

Audio Compression

Lossless compression

Generally speaking, the meaning of lossless compression is as follows: some pattern is found in the original data, and taking this pattern into account, a second stream is generated, uniquely describing the original. For example, to encode binary sequences with many zeros and few ones, we can use the following replacement:

00> 0
01> 10
10> 110
11> 111

In this case, sixteen bits:

00 01 00 00 11 10 00 00

will be converted to thirteen bits:

0 10 0 0 111 110 0 0

If we write a compressed string without spaces, we can still add spaces in it, which means restoring the original sequence.

FLAC (Free Lossless Audio Codec – Free Lossless Audio Codec)
Coding principle: the algorithm tries to describe the signal with this function so that the result obtained after subtracting it from the original (called difference, remainder, error) can be encoded with the minimum of bits.

When the model is fitted, the algorithm subtracts the approximation from the original to obtain a residual signal (error), which is then losslessly encoded.

Lossy compression (MP3, AAC, WMA, OGG)
Using a lossy compression algorithm, the size of an MP3 file with an average bit rate of 128 kbps is approximately 1/11 of the original file of an Audio CD (uncompressed audio in CD-Audio format has a rate 1411.2 kbps bit rate). MP3 files can be created at high or low bit rates, which affects the quality of the result.

The principle of compression is to reduce the precision of some parts of the sound flow, which is almost indistinguishable for most people. The audio signal is divided into segments of equal length, each of which, after processing, is packed into its own frame (frame). Spectral decomposition requires continuity of the input signal; therefore, the previous and next tables are also used for calculations. The audio signal contains harmonics with a lower amplitude and harmonics that are close to the strongest; Such harmonics are cut off, as the average human ear will not always be able to determine the presence or absence of such harmonics. This characteristic of hearing is called the masking effect. It is also possible to replace two or more close peaks with an averaged one (which, as a rule, leads to sound distortion). The cutoff criterion is determined by the outflow requirement. Since the entire spectrum is relevant, the high frequency harmonics are not cut off, but are only selectively removed to reduce information flow due to rarefaction of the spectrum. After spectral removal, mathematical compression and frame packing methods are applied.

Masking effect
In certain cases, a sound can be hidden by another sound. For example, talking next to a train track can be completely impossible if a train passes. This type of effect is called masking. A weak sound is said to be masked if it becomes indistinguishable in the presence of a louder sound.

Simultaneous masking
Any two sounds, when heard simultaneously, have an impact on the perception of the relative volume between them. A louder sound reduces the perception of a weaker one, until the disappearance of your hearing. The closer the frequency of the masked sound is to the frequency of the masker, the more it will be hidden. The masking effect is not the same when the masked sound is shifted down or up in frequency relative to masking. Low-frequency sound masks high-frequency sound. However, it is important to note that high-frequency sounds cannot mask low-frequency sounds.

Time masking
This phenomenon is similar to frequency masking, but time masking occurs here. When the masking sound is stopped, the masking remains inaudible for some time. Under normal conditions, the effect of temporary masking lasts much less. The masking time depends on the frequency and amplitude of the signal and can be up to 100 ms.
In the case where the masking tone appears later than the masking, the effect is called post-masking. When the masking tone appears before the masking (this is also possible), the effect is called premasking.

Post-stimulus fatigue
Often, after exposure to loud, high-intensity sounds, a person’s hearing sensitivity drops dramatically. Recovery of normal thresholds can take up to 16 hours. This process is called “temporary change in hearing threshold.”

Digital audio compression

Digital audio compression

Digital Audio Compression

The concept of loudness is close and understandable not only for a musician, but also for people who are not associated with music. The relationship between the volume of the parts of a piece and the volume of the instruments that are playing simultaneously is called the dynamic range. One of the main tools producers and musicians use to influence dynamic range is the compressor.

Digital Audio Compression

Although the compressor works with a known phenomenon, loudness, in most cases its use occurs spontaneously, randomly, without understanding the essence of what is happening. You can know the general principle of the compressor and the purpose of each handle, but this does not eliminate the stupor at the first experience.

Why do you need a compressor?

The main purpose of the compressor is to automatically change the signal level. It works roughly the same as if you kept your hand constantly on the volume fader, turning it up and down. The difference is that a compressor can react very quickly to changes, much faster and more accurately than a human.

Up to this point, the word compressor meant a whole class of dynamic devices. Using the same basic principles as a conventional compressor, various instruments work for different purposes: limiters, expanders, gates, etc. They are united by working with the volume of individual sounds or the mix as a whole.

The classic compressor is controversial by its very name. Everyone knows that he makes the loudest sound. But the name comes from compress, which means “compression”, and if you ask any sound engineer what a compressor does, you’ll hear the answer: “squash the signal.” The compressor reduces the amplitude of the dynamic bursts, makes them quieter. So what is the main purpose of the compressor: to make it quieter or louder? The answer is both at the same time.

Let’s take an example of voice recording. Very often, in the process of singing, syllables or sounds of different volume are heard. If the singer does not control the dynamics of his performance very well, then such differences create problems for the sound engineer and negatively affect the final result of the work. Silent syllables disappear into the mix, text becomes difficult to distinguish, and if you adjust the volume for a quiet area, in other places the voice begins to “stand out.”

This is where the compressor comes in. It allows you to suppress strong bursts, equalize them with silent fragments. Now you can turn up the volume of the track without fear of some syllables sticking out. So the compressor makes the sound lower and higher at the same time. Three images show the stages of working with sound: a source with large peaks (a), a compressed signal (b) and an increase in the volume level of the entire file (c).

It is especially important to apply compression when recording in a digital environment, when we are forced to adhere to a maximum level of 0 dB, because exceeding this threshold leads to clips and distortion. When clips appear, we lower the preamp level, which means we lower the volume of not only bursts, but quiet areas as well, leading to signal degradation due to quantization and aliasing noise.

The compressor, positioned between the preamp and the digital recording system, operates only on the loudest bursts, reducing their volume and ensuring a smooth soundtrack. Thanks to this, we have the opportunity not to reduce the overall volume of the recorded signal and to maintain the sound quality.

Unfortunately, many modern musicians, without going into the technical characteristics of the compressor, use it everywhere, believing that with its help you can “stretch” any sound in the mix. Also, compressors are often included on the road in extreme conditions. They are only used by experienced sound engineers when there is a real need.

The compressor helps avoid recording problems. The most common causes of problems can be the following:

Non-professionalism of the interpreter (dynamic unevenness).
Mismatched path (bad, mismatched, or inadequate microphones, preamps).
Disadvantages of the digital environment (limited to 0 dB).
Uncomfortable conditions for the singer (small and stuffy room, poor monitoring).
Low qualification of a recording engineer.
If a performer has a voice and can sing into a microphone, and a recording engineer knows her job well and knows how to properly position microphones and set up equipment, a compressor may not be required at all. But this is the ideal situation.

Digital audio compression

Digital audio compression

Digital Audio Compression

Audio data compression is a real problem today. There are two reasons for the need to compress audio data: memory savings when storing audio information, low bandwidth of remote digital information transmission channels. Compression effectively solves the two problems above. Data compression is an algorithmic transformation of data performed to reduce its volume.

Data Compression

It is used for a more rational use of data storage and transmission devices. Compression is based on eliminating the redundancy contained in the original data. To guarantee the parameters necessary for the transmission of voice signals (music) over modern low-speed digital communication channels and to guarantee the specified noise immunity, it is necessary to use highly efficient data compression algorithms. The transmission channel is characterized by a concept such as the capacity of the channel: And the signal – by the volume (signal): …

Both of the above features include dynamic range D, channel width (signal spectrum), and transit time T. Digital audio compressors are used to reduce dynamic range. To improve spectral efficiency, digital filters are used to limit the spectrum of the encoder output signal (according to Nyquist criteria). Among other things, encoders based on the principles of elimination of redundancy (Huffman codes) are used to guarantee a certain information transmission speed. The essence of which is as follows: codes based on the principle of assigning more probable values of the amplitudes of the codewords of shorter length than the improbable ones.

Let’s consider how the types of redundancy described above are eliminated.
Structure of a lossy audio compression encoder The original digital audio signal is divided into frequency subbands and time-segmented into a time-frequency segmentation block. The length of the encoded sample depends on the shape of the temporal function of the audio signal. In the absence of sharp peaks in amplitude, a long sample is used, which provides high-frequency resolution. In the case of abrupt changes in signal amplitude, the length of the encoded sample decreases dramatically, giving a higher time resolution. The decision to change the length of the coded sample is made by the psychoacoustic analysis unit, calculating the value of the psychoacoustic entropy of the signal.
After segmentation, the frequency subband signals are normalized, quantized, and encoded. In the most efficient compression algorithms, it is not the samples of the audio signal that are encoded, but the corresponding MDCT coefficients. (the differential between the coefficients is smaller) The accounting of the auditory perception patterns of a sound signal is carried out in the psychoacoustic analysis unit. Here, according to a special procedure, for each frequency sub-band, the maximum allowable level of quantization distortion (noise) is calculated, in which they are still masked by the useful signal of this sub-band.

The block of dynamic distribution of bits according to the requirements of the psychoacoustic model for each coding subband selects a minimum possible number of them, in which the level of distortions caused by quantization does not exceed the threshold of their audibility calculated by the model psychoacoustic.

This article will consider the functional diagrams of the audio data compression algorithms, based on µ-laws, A. The functional diagram of the compression algorithm based on the A-level compression law is shown in Fig.2. Figure 2. Functional diagram of the compression algorithm based on the A-level compression law A signal (discrete sine) is applied to the input of the compressor. After compression, the signal passes to the adder, where the noise is fed to the second input of the adder, thus simulating the additive noise of the transmission channel.

Then the noisy signal enters the input of the expander, at the output we get the reconstructed signal. The reconstructed and original signal is then fed to the adder, after which the power of the spectral noise is observed.

Simulation results (A = 87.6)
The following graphs are presented: 1-original signal, 2-signal passed through the compressor, 3-recovered signal, 4-noise power at the output of the noise generator, 5-noise power after the expander.

Improved efficiency of digital audio data compression algorithms.

Improved efficiency of digital audio data compression algorithms.

audio compression

The relevance of the work. Methods for encoding high quality (HS) audio signals have become very widespread in the last decade in the field of broadcasting, digital sound recording, and home audio and video equipment. There’s even a fast-growing new class of consumer electronics: portable MP3 players.

Audio Compression:

Digital television and radio transmission networks are being developed, providing consumers with high-quality images and sound with a wide coverage area. The popularity of radio and television broadcasts over the Internet and mobile phone networks is increasing. All these technological innovations have become economically viable, and in some cases even technically possible, thanks to the use of highly efficient digital video and audio data compression algorithms, such as MPEG-1 ISO / IEC 11172, MPEG-2 TSO / IEC 13818, MPEG-4 ISO / IEC FCD 14496, ATSC Dolby AC-3. At the same time, due to the economic advantages of using these algorithms, which make it possible to reduce the bandwidth requirements of the transmission channels or the capacity of the information carriers by an order of magnitude, it is necessary to compensate with a certain decrease in the sound quality. During the era of the dominance of digital audio CDs, consumers have created a requirement for high sound quality from any sound reproduction equipment. The efforts of algorithm developers for encoding audio signals have always been aimed at ensuring that the quality of decoded audio material is no worse than that of a CD. Sound quality is often the determining factor in the economic success of digital broadcasting services or digital sound distribution services like iTunes). Further,

It is obvious that the problem of improving the quality of audio coding is today one of the key problems for the sound recording industry, the audio broadcasting industry and the manufacturers of various multimedia systems.

The basic principle of operation of highly efficient audio coding systems is to use the properties of the human auditory system, mainly the phenomenon of masking. The phenomenon of psychoacoustic masking is due to the biophysical and neuronal processing of sound signals by the human auditory system [173]. At the same time, part of the sound information does not affect the acoustic perception of the sound signal due to the presence of components with greater intensity in it. Therefore, the strongest components of the audio signal form the so-called masking thresholds. Sound information with a signal energy level below the masking threshold is not perceived by the auditory system. In the traditional digital representation of audio signals using pulse code modulation (PCM), time-sampled samples of the original signal are represented using a specific number of bits in the code word. The finite precision of the instantaneous values of a continuous analog signal introduces an error in the signal, the so-called quantization noise. The idea of encoding audio signals with the elimination of psychoacoustic redundancy is to combine psychoacoustic analysis and the quantization mechanism of audio signals [112]. In this case, the digitally encoded signal is converted into a time-frequency representation, as close as possible to the time-frequency resolution of the human auditory system. Psychoacoustic analysis determines the masking thresholds at each point in the time-frequency representation of the encoded signal, and the quantizer re-quantizes the signal with the minimum possible number of bits per sample, in which the increasing quantization noise is still below the masking thresholds. Thus, a compact representation of audio signals can be achieved without subjective degradation of sound quality. It is obvious that the efficiency and quality of such systems depend mainly on the precision of the psychoacoustic analysis. a compact representation of audio signals can be achieved without subjective degradation of sound quality. It is obvious that the efficiency and quality of such systems depend mainly on the precision of the psychoacoustic analysis. a compact representation of audio signals can be achieved without subjective degradation of sound quality. It is obvious that the efficiency and quality of such systems depend mainly on the precision of the psychoacoustic analysis.

LEARN HOW AUDIO DATA COMPRESSION WORKS

LEARN HOW AUDIO DATA COMPRESSION WORKS

Audio Data Compression

MP3s Around Us Many, many years ago, the Internet was supposed to be the force that would democratize the music industry, physical distribution was supposed to become obsolete, and it was possible to publish music on the Internet and be heard by millions of audiences.

Audio Data Compression

In fact, enthusiasts and companies have created websites where fans can listen to new tunes, the MP3 format has made it easy to place songs for critics, and music demo pieces are now helping to sell a CD or LP. physical. It is not difficult to put your music on the Internet, but if you are not a star of the first magnitude, you will have to accept the placement of the data in compressed format to save space on the server, as well as save download time for those who download your masterpiece. While there are many critics of MP3, there are ways around some of the limitations of this format.

The MP3 format is based on the use of data compression algorithms that can reduce the amount of data required to play music. Compression algorithms in MP3 work with loss of data, they do not work like Zip or Rar compression algorithms that restore original file without data loss. MP3 algorithms discard “unnecessary” data. For example, if there is a lot of high-level sound on a track, the algorithm may assume that you cannot hear low-level material and think that only 24 dB of dynamic range is sufficient for that part of the audio material. It only requires 4 bits of data, a quarter of the data needed for 16-bit resolution. Unfortunately, it is difficult to preserve the sound quality of music when compressed, but it is possible. One way is to use algorithms, working without data loss, such as FLAC, or some algorithms offered by Microsoft and Apple for their audio formats. However, these algorithms do not lead to a significant reduction in file size; with complex music, the size reduction can be only 10-20%.

Although there are many algorithms for compressing audio data, only a few are the most common:

MP3. This format allows multiple levels of encoding, you can create audio files of almost any size with a smaller size with greater loss of precision. There are many free and shareware MP3 players (such as iTunes and Windows Media Player), to encode MP3, you can use iTunes and most digital audio editors.

AAC. As the native iPod format, this format is quite popular and sounds better than MP3 for the same file size according to most users. ITunes can convert files to AAC.

Windows Media Audio. The format is promoted by Windows, but is used less frequently than MP3 or AAC. WMA sound quality is generally better than MP3. While Microsoft does not offer users WMA playback software for the Mac platform, the Flip4Mac utility (free version available) can play Windows Media formats on Mac.

Ogg Vorbis. A great but rarely used format that sounds better than MP3 at the same bit rate, and unlike MP3, the encoding tools are free for developers. Ogg Vorbis files are not widely used yet, but they are popular with advanced technical users.

FLAC. This popular lossless format is not supported by many portable music players, but musicians often use FLAC to exchange files when working on collaborative projects. High sound quality is maintained.

Although MP3 does not offer the best quality, this format is most often used when placing audio files on the network. all players can play MP3. It is important to choose the correct MP3 settings. When encoding files to MP3, it is always best to use a high-quality source file without compression. Then select the compression settings. When saving in MP3 format, you can generally choose from a range of bit rates (bits per second), from 320 kbps stereo (great quality, but also a fairly large file) to 8 kbps mono (good enough for dictation) . In addition to the fixed settings, there is variable bit rate (VBR) encoding, which optimizes the bit stream according to the playback material. VBR encoding is not supported by all players.