Data compression


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Data compression

Data compression
Data compression

The process of encoding information using fewer bits than the original representation

Data compression
Data compression

In computer science and information theory, data compression or source coding is the process of representing information with fewer data bits (or other information-related units) than if it were not encoded, according to an encoding mechanism specific . For example, if we encode “compression” as “comp”, the item can be represented with fewer data bits. A common example is the ZIP archive format, which not only provides compression but also acts as an archiver, capable of storing many files in the same archive.

We can use data consistency (represented by information entropy, entropy), regularity, and predictability to achieve data compression. The compression technology first developed by humans is actually natural language. Generally speaking, if a thing can be described in a relatively simplified natural language, then it will be better able to compress such things. The more consistent the data, the more concentrated its statistical features. Take image compression as an example, which centrally accounts for the time domain and frequency domain of the Fourier transform, the histogram, and the eigenvalues.

 

Data compression is possible because most real-world data has statistical redundancy. For example, the letter “e” is more commonly used in English than the letter “z”, and it is very unlikely that the letter “q” will be followed by a “z”. Non-destructive data compression generally exploits statistical redundancy so that the sender’s data can be represented more succinctly, but fully.

The compression ratio of non-destructive data compression is not sufficient to handle the large volume of audio and video data, but if some loss of fidelity is allowed, higher compression can be achieved. For example, when people look at photographs or television images, they may not realize that some details are not perfect. Similarly, two audio recording sample streams may sound the same, but they are not actually exactly the same. Destructive data compression uses fewer bits to represent images, video, or audio with acceptable or imperceptible numbers.

However, there are often files that cannot be compressed using destructive data compression, and in fact cannot be compressed using any compression algorithm for data that does not contain discernible patterns. Also, trying to compress already compressed data often results in data bloat.

In fact, destructive data compression will eventually get to the point where it won’t work. For example, an extreme example: the compression algorithm deletes the last byte of the file every time, and after this algorithm continues to compress until the file is empty, the compression algorithm will not continue to work.

Compression is important because it helps reduce the consumption of expensive resources such as hard drive space and connection bandwidth, however, compression requires information processing resources, which can also be expensive. Therefore, the design of the data compression mechanism requires a compromise between the compression capacity, the degree of distortion, the computing resources required, and various other factors that must be taken into account.

As with any form of communication, compressed data communication only works if both the sender and receiver of the information understand the encryption mechanism. For example, the article only makes sense if the recipient knows that the article is to be interpreted in Chinese characters. Also, the compressed data can only be understood by the receiver if he knows the encoding method.


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Audio and video data compression

Audio and video data compression

Audio and video data compression

In computer science and information theory, data compression or source coding is the process of representing information with fewer data bits (or other information-related units) than if it were not encoded, according to an encoding mechanism specific .

Audio and video data compression

For example, if we encode “compression” as “comp”, the item can be represented with fewer data bits. A common example is the ZIP archive format, which not only provides compression but also acts as an archiver, capable of storing many files in the same archive.

We can use data consistency (represented by information entropy, entropy), regularity, and predictability to achieve data compression. The compression technology first developed by humans is actually natural language. Generally speaking, if a thing can be described in a relatively simplified natural language, then it will be better able to compress such things.

The more consistent the data, the more concentrated its statistical features. Taking image compression as an example, it centrally represents the time domain and frequency domain of the Fourier transform, the histogram, and the eigenvalues.

Data compression is possible because most real-world data has statistical redundancy. For example, the letter “e” is more commonly used in English than the letter “z”, and it is very unlikely that the letter “q” will be followed by a “z”. Non-destructive data compression generally exploits statistical redundancy so that the sender’s data can be represented more succinctly, but fully.

The compression ratio of non-destructive data compression is not sufficient to handle large volumes of audio and video data, but higher compression can be achieved if some loss of fidelity is tolerated. For example, when people look at photographs or television images, they may not realize that some details are not perfect. Similarly, two audio recording sample streams may sound the same, but they are not actually exactly the same. Destructive data compression uses fewer bits to represent images, video, or audio with acceptable or imperceptible numbers.

However, there are often files that cannot be compressed using destructive data compression, and in fact cannot be compressed using any compression algorithm for data that does not contain discernible patterns. Also, trying to compress already compressed data often results in data bloat.

What audio formats are compatible with iPhone, iPad, iPod?

What audio formats are compatible with iPhone, iPad, iPod?

Iphone

Now there are far fewer such questions on the net, but before in many forums people asked before buying an iPhone: “What audio formats are compatible with iPhone, iPad, iPod?”

Apple Music

iPhone and iPad support the following audio file formats:
AAC (8 to 320 kbps), AAC (from iTunes Store), HE-AAC, MP3 (8 to 320 kbps), MP3 VBR, Audible (formats 2, 3, 4, Audible Enhanced Audio, AAX and AAX +) , AIFF and WAV, Apple Lossless (ALAC).

Most of the time iPhone and iPad users prefer MP3 and ALAC (Apple Lossless) formats, which they download from trackers, so there is practically no problem to copy music to iPhone, iPad.

What is Apple Lossless (ALAC) and how is it different from FLAC?
A few separate words should be said about the rather unusual Apple Lossless (ALAC) – this is an analog of the FLAC audio codec. Apple Lossless was specially designed by Apple to ensure that the user can enjoy the highest quality music while keeping battery consumption within reasonable limits.

Apple Lossless (ALAC) does not require high performance, so you can listen to music without quality loss, even on old iPod Nano. Apple takes great care to ensure that its devices can work for a long time without recharging, which is why we have a FLAC analog in the person of Apple Lossless.

In what format is it better to listen to music? PART 4

In what format is it better to listen to music? PART 4

Audio File Format

What has changed today

AUDIO FORMATS

A rare sound engineer makes a digital master recording (which is then played back on physical media), using modern technologies to the full. So the chance that a 24-bit track is actually only 16-bit is extremely high.

High-quality analog recording on high-end gear is even harder to find today, if only for fans of this sound. Such is, for example, Jack White, the former leader of the White Stripes. At the same time, some of his recordings reference lo-fi variations, and looking for the scandalous sonic characteristics of the song becomes something of a foodie treat.

If you imagine an ideal source, only the trained ear or listening on high-quality audio equipment will allow you to find a compressed file. And already based on this (and without forgetting perception), it is worth drawing the following conclusion:

AAC is necessary and sufficient for medium-priced equipment, in the absence of which (and in the absence of sources that can be encoded in AAC) – MP3 with a constant 320 kbps bit rate, created with the Lame 3.93 codec (recommended keys for decoding: -cbr -b320 -q0 -k -ms).

The exceptions are recordings originally recorded in high quality, say, recorded on DVD-Audio, SACD, or recordings originally collected in DSD (or similar format) with a high bit rate.

Although without losses it has some characteristics. And we will tell about them next time.

The author does not like Apple. The author greatly appreciates the achievements of the Fraunhofers and was greatly surprised to learn that AAC is his work. 🙂

In what format is it better to listen to music? Part 3

In what format is it better to listen to music? Part 3

audio formats

Due to its advanced age, MP3 has significant limitations: the bit depth can be 16-24 bits, the sample rate is expressed only in discrete values ​​(8, 11,025, 12, 16, 22.05, 24, 32, 44.1, 48), the bit rate is limited to 320 kbps. Also, in the normal version of MP3, the number of channels is limited to two.

audio formats

AAC
The same rake, only in profile. Also developed by the Fraunhofer Society. Later and uses a different, more modern psychoacoustic model. The publicly available information allows us to conclude: yes, they managed to improve their own creation.

Even with the simplest numbers, AAC is a more flexible format. The bit depth of the files obtained with the help of this development varies from 16 to 24, the sampling frequency, if desired, will also allow not to lose the sound image and is in the range of 8-192 kHz. The data stream is generally close to lossless formats (up to 512 kbps), while the maximum number of AAC file channels reaches 48.

Which format is definitely the best?
Considering that AAC is MP3 reinvented after a dozen years, then the choice is in its favor. If you want, it makes sense to only compare MP3 and OGG.

On the graphics – good AudioCD, compressed OGG with 350 kbps variable bit rate and MP3 using Lame. The lower the graph, the closer the sound is to the original. It turns out to be a very interesting image. Although MP3 has clearly cut the high frequencies, unlike OGG, in which you can see a blockage below 2 kHz.

The frequency-time distribution of sound does not speak of less interesting things. At a constant 320kbps bit rate, MP3 is almost identical to the original recording. Everything seems to fit now. But … In fact, everything is even more confusing.

Why use at a loss at all when there is no loss available?
Common sense.

The fact is that most analog recordings do not contain the amount of information that would need to be stored in high-quality formats. Don’t forget that the native sample rate for CD is 44.1 kHz, the quantization is only 16 bits.

The above graphics well demonstrate the high fidelity of MP3 streaming. But for an audio cassette, magnetic tape (unless of course it is a master tape), the characteristics of an audio CD are unattainable. And for mass studio equipment, the ability to record analog sound corresponding to AudioCD has appeared relatively recently. It makes no sense to digitize in FLAC (and even more so in WAV) a concert recording or a disc from the pre-digital era, especially those made with magnetic media. They do not contain those spectra and the amount of information that containers can store without compression.

In what format is it better to listen to music? Part 2

In what format is it better to listen to music? Part 2

Audio Formats

The reference value of the audible range for humans is 16 Hz to 20 kHz, but you cannot hear and be aware of all incoming sounds simultaneously.

audio files

Hearing is discreet and your hearing sensitivity is not linear.

Modern psychoacoustic models accurately assess human hearing and are constantly improving. In fact, despite the guarantees of music lovers, musicians and audiophiles, to the inexperienced middle ear, the initial appearance of MP3 in maximum quality has become extremely noticeable. There are exceptions, they cannot cease to exist. But they are not always easily noticed by blind listening.

Formats using psychoacoustic compression models
There are many of these formats for lossy audio compression. The most common today are the following.

OGG (Vorbis)
In general, a file with the * .ogg extension is a “container”: it can contain multiple sound recordings with their own tags and characteristics. Most of the time, the files stored in it are compressed with the Ogg Vorbis codec, although others can be used, including MP3 or FLAC.

Its main advantages include a wide range of possible parameters during encoding: the audio sampling frequency can reach 192 kHz, the bit depth is 32 bits. By default, OGG uses a variable bit rate (although this is not shown on the properties screen), which can go up to 1000 kbps.

MP3
Unlike the free OGG, MP3 was developed by the Fraunhofer Society, an association of German institutes for applied research, which is very important for modern acoustics. Among audiophiles, by the way, this is an extremely respected office, yet they don’t like to admit it. But its developments are closely watched.

Unlike OGG, it can have variable (VBR) and constant (CBR) bit rate. By the way, it was thanks to MP3 that it was discovered that not all recordings can be encoded with high quality with a variable bit rate (see the above reasons, the encoding algorithms and their results in this case may be different when encoding the same source ).

In what format is it better to listen to music?

In what format is it better to listen to music?

Lossy compression

Understanding digital audio formats is not easy. It is even more difficult to come to an unequivocal conclusion in which format it is better to listen to music.

Lossy Formats

If you look at the audio format comparison table on Wikipedia, your eyes will start to flutter with columns of silent numbers. Let’s try to find out what’s behind this.
In what format is it better to listen to music? Three lost whales
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Let’s make a reservation right away that the article talks ONLY about general characteristics and will not include some details. Moving forward, Lifehacker will conduct its own unbiased investigation. And today we will try to generalize the already known experience in one way or another.

There is an analog and a figure.

The analog is good, but short-lived and inconvenient. Therefore, analog media, despite high vinyl sales, will not be making a comeback.

Digital audio can be of three main types:

in a format that does not use compression;
in a format that uses lossless compression;
in a format that uses lossy compression.
At first glance, lossless formats are more promising. This is not always the case, as we will discuss in more detail in one of the following materials. Uncompressed formats make no sense other than storing the master recordings needed to create audio content. They are easier to restore. Storing and listening to home recordings is superfluous.

Of the many parameters of digital audio, the user must first be concerned with sample rate (the accuracy of digitizing an analog signal in time), bit depth (the accuracy of digitizing in amplitude – volume) , the bit rate (the amount of information contained in the file in terms of one second).

Today we will talk about lossy.

For compressed sound, the concept of the psychoacoustic model is very important – the ideas of scientists and engineers about how a person perceives sound. The ear perceives the entire spectrum of acoustic waves entering it. However, the brain processes the signals.

Lossy compression format at a glance

Lossy compression format at a glance

lossy compression

“As you know, the music we listen to consists of a set of signals, each of which has its own characteristics, including loudness.

LOSSY COMPRESSION

The human hearing aid is designed so that we do not distinguish or poorly distinguish a weak (low) signal from the background of a strong (strong) signal. This principle forms the basis of modern means of compression (compression) of audio data.

If we imagine that a signal of a certain length is divided into many parts, and each part is processed in such a way that a weaker signal, which is difficult to distinguish from the background of a strong one, falls under the knife, and one remains a signal louder, then this will be an approximate audio compression model. Consequently, the level of data compression will depend on how many parts (samples) the original file will be divided into and how many weak signals from each individual sample will be removed (what the bit rate will be, the number of bits in a sample). sample of a specified duration). This coding principle is called lossy coding or lossy coding.

Ogg Vorbis is a completely open and patent-free audio format that allows you to store and transmit audio information with high sound quality (44.1-48.0 kHz sample rate, more than 16 bits, polyphony (multi-channel audio) ) and bit rates ranging from 16 to 512 Kbps per channel. In this case, the number of processed channels can reach 255.

MP3 or MPEG-1 Layer 3 audio is by far the most popular format for storing and transmitting compressed data. This format was developed by the Fraunhofer Institut, Germany. “Http://ru.wikibooks.org/wiki/Compression_Audio_data_with_lossy

Comparative tests

Sound Forge 7.0 (Spectral Analysis / Spectrum Analysis function) was used for the analysis of the sound signal.

“Spectral analysis is a signal processing technique that can reveal the frequency content of a signal. Solving the problems of spectral analysis is possible through the use of the fast Fourier transform, which makes it possible to determine the contribution of individual components of the vibration spectrum to the overall vibration picture. “Http: //masters.donntu. edu.ua/2007/fema/belinskaya/library/a4/art4.htm

The following graphs were obtained in the form of an amplitude distribution in the frequency domain, the spectrum of the signal is presented using a Blackman-Harris / Blackman-Harris window and a maximum sampling frequency (FFT size) of 65536, this gives allows you to analyze the smallest details of the signal at frequencies around 20,000 Hz, without smoothing.

The analysis of the spectrum of the compressed signal assumes the presence of a recording of the original quality, for this we use a licensed audio CD made in the USA “Kevin Yost – Bongo Madness”, with standard characteristics 44100 Hz / 16 bit

The rich electronic sound spans the entire frequency spectrum and captures even the inaudible range (20,000 Hz to 22,000 Hz), as can be seen in the graph below. Considering that it is generally possible to notice codec compression at higher frequencies, the 10-20 kHz range will be considered.