Tag: what digital audio is

Digital audio information (Part 3)

Codec sample rate and bit depth

Sampling is the acquisition of instantaneous values ​​(samples) of an analog signal with a certain time step in the digitization process. The frequency of this step is called the sample rate (it is also the sample or sample rate). The larger it is, the better the sound recorded and reproduced. In studio equipment, the frequency is 48 kHz, in home systems – 44.1 kHz.

Bit depth determines the quality of the recorded audio. Higher is better. The bit value, for example 32, denotes the number of bits that are allocated to record the amplitude of the signal at the time of its measurement.

Consequently, the more often (sample rate) and more accurately (bit depth) the audio signal is measured, the higher quality audio file is obtained.

Bitrate

The bit rate (literally, the information bit rate) determines the maximum amount of information that can be transmitted through the audio channel per unit of time. A high bit rate is needed to transmit a rich sound image and is not required when encoding speech. Audio recordings with a 128 Kbps bit rate are suitable for inexpensive speakers, but when accessing expensive equipment, it makes sense to get music at a 192-256 Kbps bit rate.

Convenient solution: variable bit rate encoding, change the bandwidth of the audio channel according to the quality and saturation of the musical fragment.

Audio formats

MP3 is the most popular digital audio format right now. It is widely used in file-sharing networks due to the small size of the final files (approximately 1/10 of the original audio CD file) and due to its special data compression algorithm, it provides playback quality very close to that of original. The MP3 format is compatible with absolutely all RoverMedia players, as well as all modern stereos and DVD players.

WMA is a file format developed by Microsoft to store and transmit audio information. The main advantage of WMA over MP3 is its greater compression capacity, which results in a smaller file size. The latest versions of the format, starting with Windows Media Audio 9.1, provide lossless encoding, multi-channel surround sound encoding, and speech encoding.

WAV is an audio container file format for storing a recording of a digitized audio stream. This format is mainly used to record sound from the voice recorder built into RoverMedia players and most modern devices.

FLAC (Free Lossless Audio Codec) is one of the most popular formats for lossless audio compression. Unlike MP3 and WMA formats, it does not remove any information from the audio stream when encoding the audio. Thanks to this, FLAC files are suitable not only for listening to high-quality music on RoverMedia portable media players, but even on high-quality audio equipment.

Number of audio channels

Infectious mononucleosis
Mono (from the Greek (Monos) – one) is a prefix that means the relationship with the singular.
Mono eng. Mono (monophony) is most often used as a term related to the recording and reproduction of sound.
Mono means monophonic, single channel.

Stereo
Stereo (from Greek solid, spatial)
Stereophony or stereo sound (from the ancient Greek words “stereoros” – solid, spatial and “background” – sound): recording, transmission or reproduction of sound, in which the auditory information about the location of its source is stored through sound design over two (or more) independent audio channels. …
In stereo recording, the recording is made from 2 microphones spaced a certain distance, each with a separate channel (right or left).
The result is what is called “panoramic sound”.

Digital audio information (Part 2)

Sampling frequency. A microphone connected to the sound card is used to record analog sound and convert it to digital format. The quality of the digital sound obtained depends on the number of measurements of the sound volume level per unit of time, that is, the sampling frequency. The more measurements that are made in 1 second (the higher the sampling frequency), the more accurately the “ladder” of the digital audio signal repeats the curve of the dialogue signal.

The audio sample rate is the number of sound volume measurements in one second.

The audio sample rate can vary between 8000 and 48000 sound volume measurements per second.

Audio encoding depth. Each “step” is assigned a specific value for the sound volume level. Loudness levels of sound can be viewed as a set of possible states N, for which a certain amount of information I is required, which is called audio coding depth.

Audio encoding depth is the amount of information required to encode the discrete volume levels of digital audio.

If the encoding depth is known, then the number of digital audio loudness levels can be calculated using the formula N = 2I. Let the sound encoding depth be 16 bit, then the number of sound volume levels is:

N = 2I = 216 = 65536.

During the encoding process, each sound volume level is assigned its own 16-bit binary code, the lowest sound level will correspond to the code 0000000000000000 and the highest – 1111111111111111.

The quality of digitized sound. The higher the sampling frequency and depth of the sound, the better the sound of the digitized sound. The lowest quality of digitized sound, corresponding to the quality of telephone communication, is obtained at a sampling rate of 8000 times per second, a sampling rate of 8 bits, and by recording an audio track (“mono” mode). The highest quality of digitized sound, corresponding to the quality of an audio CD, is achieved with a sampling rate of 48,000 times per second, a sampling rate of 16 bits and the recording of two audio tracks (stereo mode) .

It should be remembered that the higher the quality of the digital sound, the greater the volume of information in the audio file. It is possible to estimate the volume of information of a digital stereo sound file with a duration of 1 second with an average sound quality (16 bits, 24,000 measurements per second). To do this, the encoding depth must be multiplied by the number of measurements in 1 second and multiplied by 2 (stereo sound):

16 bit? 24,000? 2 = 768,000 bits = 96,000 bytes = 93.75 KB.

Sound editors. Sound editors allow you not only to record and play sound, but also to edit it. Digitized sound is presented in sound editors visually, so copying, moving, and deleting parts of the audio track can be easily performed with the mouse. Furthermore, you can layer audio tracks on top of each other (mix sounds) and apply various acoustic effects (echo, reverse playback, etc.).

Sound editors allow you to change the digital sound quality and volume of an audio file by changing the sample rate and encoding depth. Digitized audio can be saved uncompressed as universal WAV or compressed MP3 audio files.

By storing audio in compressed formats, low-intensity audio frequencies “excessive” for human perception are discarded, coinciding in time with high-intensity audio frequencies. Using this format allows you to compress audio files dozens of times, but it leads to irreversible loss of information (files cannot be restored in their original form).
test questions

1. How do sample rate and encoding depth affect digital audio quality?
Self-help assignments

1.22. Selective Response Mapping. The sound card performs binary encoding of the analog audio signal. How much information is needed to encode each of the 65,536 possible levels of signal intensity?
16 bits;
256 bits;
1 bit;
8 bits.

1.23. A task with a detailed answer. Estimate the volume of information in digital audio files with a duration of 10 seconds at an encoding depth and a sample rate of an audio signal that provides the minimum and maximum sound quality:

a) mono, 8 bits, 8000 measurements per second;

b) stereo, 16 bits, 48,000 measurements per second.

Digital audio information (Part 1)

The history of recording technology

The creation of sound by computer is a modern stage in the history of the development of sound technology. Let’s take a brief look at this story.

Since the late 19th century, the technical means of storing and transmitting information have developed rapidly. So in the late 1800s, the famous American inventor Thomas Edison made a phonograph.

The principle of operation of the phonograph is as follows. Speech, music, or song create sound vibrations that are transmitted to the recording pen of the phonograph. The needle, acting on the surface of the rotating wax roller, leaves in it a groove with variable depth: a sound track. When a sound is reproduced, the opposite process occurs: the movement of the reading needle along the soundtrack is accompanied by its oscillations with the same frequency. These vibrations are converted by the phonograph into an audible sound. The Edison phonograph is the first sound recording device.

The same idea served as the basis for the production of celluloid gramophone records and mechanisms that reproduce the sound recorded on them: gramophone and gramophone.

In the middle of the 20th century, an electrophone appeared, an electrical analog of a gramophone.
Analog sound representation

The soundtrack of a phonograph record is an example of a continuous form of sound recording.

The electrical signal is transmitted to the speaker of the microphone and converted into sound.

In the 20th century, the tape recorder was invented, a device for recording sound on magnetic tape. It also uses an analog form of audio storage. Only now the soundtrack is not a mechanical “pit groove”, as shown in fig. 1.1, and a line with continuously changing magnetization. With the help of a magnetic reading head, an alternating electrical signal is generated, which is emitted by an acoustic system.

Until recently, all sound transmission technology was analog. This is both telephone communication and radio communication. During a telephone conversation, the sound vibrations from the microphone membrane are converted into an alternating electrical signal that is transmitted through electrical cables. On the receiving phone, they become sound.
Audio encoding and processing

Sound information. Sound is a wave that travels through air, water, or other medium with a continuously varying intensity and frequency.

A person perceives sound waves (air vibrations) with the help of hearing in the form of sound of different volume and pitch. The greater the intensity of the sound wave, the louder the sound, the higher the frequency of the wave, the higher the pitch of the sound.
Dependence of the volume and pitch of the sound on the intensity and frequency of the sound wave.

The human ear perceives sound at a frequency of 20 vibrations per second (low sound) to 20,000 vibrations per second (high sound).

A person can perceive sound in a wide range of intensities, in which the maximum intensity is 1014 times greater than the minimum (one hundred thousand billion times). To measure the volume of sound, a special unit “decibel” (dbl) is used (Table 5.1). A decrease or increase in sound volume by 10 dB corresponds to a decrease or increase in sound intensity by 10 times

Sound volume

Sound volume in decibels:
-Lower limit of human ear sensitivity 0
-Rustling leaves 10
-Talk 60
-90 car horn
-120 jet engine
-Pain threshold 140

Sound time sampling. (Part 1)

In order for a computer to process sound, a continuous audio signal must be converted to a discrete digital form using time sampling. A continuous sound wave is divided into separate small time sections, for each section a certain value of sound intensity is set.

Therefore, the continuous dependence of the loudness of the sound at time A (t) is replaced by a discrete sequence of loudness levels. On the graph, this appears to replace a smooth curve with a sequence of “steps”).

Parameters that affect the quality of digital audio. (Part 3)

The quality of the digital sound obtained depends on the number of measurements of the sound volume level per unit of time, that is, the sampling frequency.

Audio sample rate is the number of audio volume measurements in one second.

The more measurements that are made in one second (the higher the sampling frequency), the more accurately the “ladder” of the digital audio signal repeats the curve of the analog signal.

Each “step” of the graph is assigned a certain value for the sound volume level. Loudness levels can be thought of as a set of possible N states (gradations), which require a certain amount of I information to encode, which is called audio encoding depth.

Audio encoding depth is the amount of information required to encode the discrete volume levels of digital audio.

If the known encoding depth, the number of digital audio volume levels can be calculated by the general formula N = 2 I.

For example, if the audio encoding depth is 16-bit, then the number of audio volume levels is:

N = 2 I = 2 16 = 65 536.

During the encoding process, each sound volume level is assigned its own 16-bit binary code, the lowest sound level will correspond to the code 0000000000000000 and the highest – 1111111111111111.

Digitized audio quality

Therefore, the higher the sample rate and depth of audio encoding, the better the digitized sound will sound and the better you can bring the digitized sound closer to the original sound.

The highest quality of digitized sound, corresponding to the quality of an audio CD, is achieved with a sampling rate of 48,000 times per second, a sampling rate of 16 bits and the recording of two audio tracks (stereo mode) .

It should be remembered that the higher the quality of the digital sound, the greater the volume of information in the audio file.

You can easily estimate the volume of information in a digital stereo sound file with a duration of 1 second with an average sound quality (16 bits, 24,000 measurements per second). To do this, the encoding depth must be multiplied by the number of measurements per second and multiplied by 2 channels (stereo sound):

16 bits × 24,000 × 2 = 768,000 bits = 96,000 bytes = 93.75 KB.

There are three main types of audio digits:

format – no compression;
format (lossy) – lossy compression;
format (lossless): lossless compression.
Lossy compression: technology in which there is a significant reduction of the encoded file compared to the original, due to the removal of information that is not perceived by the human ear.

The downside of this technology is the fact that the compressed file will never be identical to the original.

List of the most common lossy formats:

AAC (.m4a, .mp4, .m4p, .aac): advanced audio encoding (often in MPEG-4 container)
MP2 (MPEG Layer 2)
MP3 (MPEG Layer 3)
MPC (known as Musepack, previously called MPEGplus or MP +)
Ogg Vorbis
WMA (Windows Media Audio)

Lossless – Lossless compressed audio formats, including:

FLAC (Free Lossless Audio Codec)
APE (mono audio)
WV (WavPack)
These formats are capable of converting CD to digital format while maintaining quality. As an example, you can take a CD, convert it to WAV, then WAV to FLAC, then go back from FLAC to WAV, and then burn it to a blank CD and you have an absolutely identical copy of your source.

What format does the music sound with the best quality?
The most popular is the lossless FLAC format, and one of the most widely used CD to FLAC conversion programs is EAC (Exact Audio Copy).

Of all the parameters of digital audio, it is necessary to pay attention primarily to the following indicators:

sampling rate (precision of digitizing an analog signal in time),
bit rate (amount of information contained in a file in terms of one second).

The sample rate is the frequency at which digital audio is processed. The most common sample rate for quality audio formats is 44.1 kHz.

It is generally accepted that a high bit rate guarantees the best quality; this is true, but only if the source file is of good quality. A high quality MP3 should have a bit rate of 320 kbps, but a high quality FLAC format generally has a bit rate of 900 kbps and higher.

What is the best quality music format?
In addition to the audio formats themselves, for high-quality music sound, high-quality reproduction equipment is also needed: speakers, amplifiers, headphones.

Parameters that affect the quality of digital audio. (Part 2)

The format is also called the number of channels in multichannel sound systems (5.1; 7.1). Initially such a system was developed for cinemas but later spread by Software Codec

 

Software-level audio codec

§ G.723.1 – one of the basic codecs for IP telephony applications

§ G.729 – proprietary narrowband codec used for digital representation of speech

§ Internet Low Bit Rate Codec (iLBC) – a popular free codec for IP telephony (in particular for Skype and Google Talk)

Audio Codec (Audio Codec; Audio Encoder / Decoder) – A computer program or hardware designed to encode or decode audio data.

Software codec

A software-level audio codec is a specialized computer program, a codec that compresses (compresses) or decompresses (decompresses) digital audio data according to an audio file format or streaming audio format. The task of an audio codec as a compressor is to provide an audio signal with a certain quality / precision and the smallest possible size. Compression reduces the amount of space required to store audio data, and it is also possible to reduce the bandwidth of the channel through which the audio data is transmitted. Most audio codecs are implemented as software libraries that interact with one or more audio players such as QuickTime Player, XMMS, Winamp, VLC media player, MPlayer, or Windows Media Player.

Popular software audio codecs by application:

§ MPEG-1 Layer III (MP3) is a proprietary audio recording codec (music, audiobooks, etc.) for computer equipment and digital players

§ Ogg Vorbis (OGG) – the second most popular format, widely used in computer games and file-sharing networks to transfer music

§ GSM-FR is the first digital voice coding standard used in GSM phones

Adaptive Multispeed (AMR): human voice recording on mobile phones and other mobile devices

Dependence of the loudness, as well as the tone of the sound on the intensity and frequency of the sound wave.

Hertz (denoted by Hz or Hz) is a unit of measurement for the frequency of periodic processes (eg, oscillations).
1 Hz means an execution of said process in one second: 1 Hz = 1 / s.

If we have 10 Hz, this means that we have ten executions of said process in one second.

The human ear can perceive sound at frequencies ranging from 20 vibrations per second (20 Hertz, low sound) to 20,000 vibrations per second (20 KHz, high sound).

In addition, a person can perceive sound in a wide range of intensities, in which the maximum intensity is 1014 times greater than the minimum (one hundred thousand billion times).

To measure the volume of sound, a special unit of “decibels” (dB) was invented and used.

A decrease or increase in sound volume by 10 dB corresponds to a decrease or increase in sound intensity by 10 times.

Sound volume in decibels

In order for computer systems to process sound, a continuous audio signal must be converted to a discrete digital form by time sampling.

For this, a continuous sound wave is divided into separate small time sections, for each section a certain value of sound intensity is set.

Therefore, the continuous dependence of the loudness of the sound at time A (t) is replaced by a discrete sequence of loudness levels. On the graph, this appears to replace a smooth curve with a sequence of “steps.”

Sync Audio Sampling

A microphone connected to the sound card is used to record analog audio and convert it to digital format.

The denser the discrete fringes are located on the graph, the better it is ultimately possible to recreate the original sound.

Parameters that affect the quality of digital audio. (Part 1)

The best music formats for sound quality Minimum and maximum sound quality

The main parameters that affect the quality of digital audio recording are:

§ The capacity of the ADC and DAC.

§ Sampling frequency of ADC and DAC.

§ Jitter ADC and DAC

§ Resampling

In addition, the parameters of the analog path of digital sound recording and playback devices are still important:

§ Signal to noise ratio

§ Total harmonic distortion

§ Intermodulation distortion

§ Inequality of the amplitude-frequency response

Channel interpenetration

§ Dynamic range

Digital sound recording techniques

Digital sound recording is currently done in recording studios, under the control of high-quality, expensive personal computers and other equipment. In addition, the concept of “home studio” is quite developed, in which professional and semi-professional recording equipment is used, allowing you to create high-quality recordings at home.

Sound cards are used as part of the computers that perform processing on your ADCs and DACs; Most of the time at 24 bit and 96 kHz, a further increase in bit rate and sample rate hardly increases the recording quality.

There is a whole class of computer programs: sound editors that allow you to work with sound:

Record incoming audio stream

§ create (generate) sound

§ modify an existing recording (add samples, change timbre, speed of sound, cut parts, etc.)

§ rewrite from one format to another

Convert convert different audio codecs

Some simple programs only allow converting formats and codecs.

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 audio formats (APE, FLAC)

Lossy compression 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 multi-channel 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.

Why is today’s audio and video called digital?

The current heyday of mobile devices, computers, and the World Wide Web is called the digital age. This is due to the rise of digital information: everything we read and write, see and hear, is translated into a form that a computer can understand. The computer, in turn, opens up a whole universe of possibilities for working with such information: it is becoming easier to copy, transfer and store melons. MTS / Media will help you understand the theory of creating a digital world.

Binary and decimal number systems

Before understanding how an image obtained by the lens of a smartphone, or a book, or is converted into a file on a computer, you must understand at least a first approximation of how this same computer works.

At the most basic level, the computer, despite the buzz attributed to intelligence, operates with absolutely primitive categories: yes, no, no, no. In the jungle of microcircuits, this dualism is expressed in the presence or absence of an electrical signal. Everything that a computer has to digest must first be “chewed”, decomposed into simple elements, reduced to a set of two opposing concepts.

“No” in computer language replaces the number 0, “yes” – 1. That is why computer information is called digital. Everything your computer or smartphone stores, all the complex algorithms built into the most complex programs, and a masterpiece frame from the last party, and your favorite song, and an unfinished letter to your boss with the title “let’s go. … “, this is all just a long string of zeros and ones.

The base number in our daily life is 10; We use numbers from 0 to 9, that is, the decimal number system is familiar to us. In the world of computers, the base number is 2 (just two digits, 0 and 1), and the number system is called binary or binary. In the decimal system, to go from single-digit numbers to two-digit numbers, you must first count to nine, to go to three-digit numbers, up to 99. The principle of digit formation in systems is the same: appears a new digit in a number after all available digits in the current one have been used up.

Now we understand how any number can be converted to a digital form, understandable to a computer. Also, we can see what it is, the minimum information is 1 or 0. This minimum piece is called a bit. To write the number 2 in the binary system, you need 2 bits of information (10), to write the number 4 – 3 bits (100), for 15 – 4 bits (1111).

Letters in numbers

In fact, most of the time we are not dealing with bits of information, but with bytes. A byte has 8 bits. If you see that we are talking about the amount of information, say 10 MB, then the letter “B” is exactly one byte, not one bit. In cases where bits are indicated, the word “bit” is written in its entirety.

A byte is an analog of a word in machine language. At the dawn of the computer age, 8 bits of information corresponded to a memory cell of machines, the 8 bits were transmitted together as a whole. Then the “words” from the machine started to get longer, but they were still multiples of eight times the number of bits.

Why exactly 8? It happened like that. 8 bits were needed to represent 1 character of text in one of the earliest computer encodings. An encoding is a table of correspondence between text characters and binary numbers. If you try to type all variants of eight-digit numbers consisting of zeros and ones, from 00000000 to 11111111, there will be 256 such variants, that is, how many characters are in many existing encodings, and they are all called 8-bit.

A coding table is a kind of instruction for a computer with which it translates the letters of a text into binary numbers and vice versa. However, not all characters from all languages ​​fit in one encoding, and each language needs its own instructions. For this reason, national encodings have become widespread in the world. So, in the Cyrillic encodings (KOI-8, Windows-1251, MacCyrillic) there are large and small letters of the Russian and Latin alphabets, numbers, punctuation marks and auxiliary symbols. If support for Cyrillic encoding is not installed on a computer somewhere in China, you will not be able to type Russian characters and the operating system will not be able to display them.

Later, along with 8-bit encodings, 16-bit encodings became widespread, in which almost every imaginable character of every language can be found. However, each letter in this encoding already has two bytes.

So 1 letter is 1 or 2 bytes.