The main parameters affecting the quality of digital audio recording are:
§ Bit capacity of ADC and DAC.
§ ADC and DAC sampling rates.
§ Jitter ADC and DAC
§ Oversampling
Also important are the parameters of the analogue path of digital sound recording and sound reproduction devices:
§ Signal to noise ratio
§ Harmonic distortion factor
§ Intermodulation distortion
§ Uneven amplitude-frequency characteristics
§ Interpenetration of channels
§ Dynamic range
Digital audio recording technology
Record digital audio currently carried out in recording studios, controlled by personal computers and other expensive and high-quality equipment. The concept of a “home studio” is also quite widely developed, in which professional and semi-professional recording equipment is used, which allows you to create high-quality recordings at home.
Apply sound cards as part of computers that perform processing in their ADCs and DACs - most often in 24 bits and 96 kHz, further increasing the bit rate and sampling frequency practically does not increase the quality of the recording.
There is a whole class of computer programs - sound editors that allow you to work with sound:
§ record incoming audio stream
§ create (generate) sound
§ change existing entry(add samples, change timbre, sound speed, cut parts, etc.)
§ rewrite from one format to another
§ convert convert different audio codecs
Some simple programs, allow only conversion of formats and codecs.
Types of digital audio formats
There are different concepts of sound format.
The format for representing audio data in digital form depends on the quantization method used by the digital-to-analog converter (DAC). In audio engineering, two types of quantization are currently most common:
§ pulse code modulation
§ sigma-delta modulation
Often, the quantization bit depth and sampling frequency are indicated for various audio recording and playback devices as the digital audio presentation format (24 bit/192 kHz; 16 bit/48 kHz).
The file format determines the structure and presentation features of audio data when stored on a PC storage device. To eliminate redundancy in audio data, audio codecs are used to compress audio data. There are three groups of sound file formats:
§ Uncompressed audio formats such as WAV, AIFF
§ audio formats with lossless compression (APE, FLAC)
§ audio formats using lossy compression (mp3, ogg)
Modular music file formats stand out. Created synthetically or from samples of pre-recorded live instruments, they mainly serve to create modern electronic music (MOD). This also includes the MIDI format, which is not a sound recording, but with the help of a sequencer it allows you to record and play music using a specific set of commands in text form.
Digital audio media formats are used both for mass distribution of sound recordings (CD, SACD) and in professional sound recording (DAT, minidisc).
For surround sound systems, it is also possible to distinguish audio formats, which are mainly multi-channel audio accompaniments for films. Such systems have entire families of formats from two large competing companies, Digital Theater Systems Inc. - DTS and Dolby Laboratories Inc. - Dolby Digital.
The format is also the number of channels in multichannel sound systems (5.1; 7.1). Initially, such a system was developed for cinemas, but was subsequently expanded Software codec
Audio codec on program level
§ G.723.1 - one of the basic codecs for IP telephony applications
§ G.729 is a proprietary narrowband codec that is used for digital representation speeches
§ Internet Low Bitrate Codec (iLBC) - a popular free codec for IP telephony (in particular, for Skype and Google Talk)
Audio codec(English) Audio codec; audio encoder/decoder) - computer program or hardware, designed to encode or decode audio data.
Software codec
Audio codec at the program level is specialized computer program, a codec that compresses (compresses) or decompresses (decompresses) digital audio data according to a file audio format or streaming audio format. The job of an audio codec as a compressor is to provide an audio signal with a specified quality/accuracy and the smallest possible size. Compression reduces the amount of space required to store audio data and can also reduce the bandwidth of the channel over which audio data is transmitted. Most audio codecs are implemented as software libraries that interface 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) - proprietary audio codec (music, audiobooks, etc.) for computer equipment and digital players
§ Ogg Vorbis (OGG) - the second most popular format, widely used in computer games and in file-sharing networks for transmitting music
§ GSM-FR - the first digital speech coding standard used in GSM phones
§ Adaptive multi rate (AMR) - human voice recording mobile phones and other mobile devices
The dependence of the volume, as well as the pitch of sound, on the intensity and frequency of the sound wave
Hertz(indicated by Hz or Hz) - a unit of measurement of the frequency of periodic processes (for example, oscillations).
1 Hz means one execution of such a process in one second: 1 Hz = 1/s.
If we have 10 Hz, then this means that we have ten executions of such a 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 over a wide range of intensities, in which the maximum intensity is 1014 times greater than the minimum (one hundred thousand billion times).
In order to measure the volume of sound, a special unit was invented and used " decibel" (dB)
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 to computer systems could process sound, the continuous audio signal must be converted into digital, discrete form using time sampling.
To do this, a continuous sound wave is divided into separate small temporary sections, and for each such section a certain value of sound intensity is set.
Thus, the continuous dependence of sound volume on time A(t) is replaced by a discrete sequence of loudness levels. On the graph, this looks like replacing a smooth curve with a sequence of “steps”.
Time sampling of audio
For recording analog sound and converting it into digital form uses a microphone connected to the sound card.
The denser the discrete stripes are located on the graph, the better quality you will eventually be able to recreate the original sound.
The quality of the resulting digital sound depends on the number of measurements of the sound volume level per unit time, i.e. the sampling frequency.
Audio sampling rate is the number of sound volume measurements in one second.
The greater the number of measurements taken in one second (the higher the sampling frequency), the more accurate the “ladder” of the digital sound signal repeats the analog signal curve.
Each “step” on the graph is assigned a specific sound volume level value. Sound volume levels can be thought of as a set of possible states N(gradations), for coding of which it is necessary a certain amount of information I, which is called the audio coding depth.
Audio coding depth is the amount of information needed to encode discrete volume levels of digital audio.
If the encoding depth is known, then the number of digital sound volume levels can be calculated using the general formula N=2I.
For example, let the audio encoding depth be 16 bits, in which case the number of audio volume levels is equal to:
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 sound quality
So, the higher the sampling frequency and the audio encoding depth, the higher quality the digitized audio will sound and the better you can bring the digitized audio closer to the original sound.
The highest quality digitized audio, corresponding to audio CD quality, is achieved with a sampling rate of 48,000 times per second, a sampling depth of 16 bits and recording of two audio tracks (stereo mode).
It must be remembered that the higher the quality of digital sound, the larger the information volume of the sound file.
You can easily estimate the information volume of a digital stereo audio file with a sound duration of 1 second with 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.
Sound editors
Sound editors allow you not only to record and play back sound, but also to edit it. The most prominent can be safely called, such as Sony Sound Forge , Adobe Audition, GoldWave and others.
Digitized sound is presented in sound editors in a clear visual form, so copying, moving and deleting parts of the audio track can be easily done using a computer mouse.
In addition, you can overlap, overlap audio tracks on each other (mix sounds) and apply various acoustic effects (echo, playback in reverse, etc.).
When saving sound in compressed formats, low-intensity sound frequencies that are inaudible and imperceptible (“excessive”) for human perception, coinciding in time with high-intensity sound frequencies, are discarded. Using this format allows you to compress sound files tens of times, but leads to irreversible loss of information (files cannot be restored in their original, original form).
There are three main types of audio figures:
- format - no compression;
- format (lossy) - lossy compression;
- format (lossless) - lossless compression.
Lossy - lossy compression: a technology that significantly reduces the encoded file in comparison with the original, due to the removal of information that is not perceptible to the human ear.
The disadvantage of this technology is the fact that compressed file will never be identical to the original.
List of the most common lossy formats:
- AAC (.m4a, .mp4, .m4p, .aac) - Advanced Audio Coding (often in an MPEG-4 container)
- MP2 (MPEG Layer 2)
- MP3 (MPEG Layer 3)
- MPC (known as Musepack, formerly known as MPEGplus or MP+)
- Ogg Vorbis
- WMA (Windows Media Audio)
| Format | Quantization, bit | Sampling frequency, kHz | Amount of data flow from disk, kbit/s | Compression/packing ratio |
|---|---|---|---|---|
| DTS | 20-24 | 48; 96 | before 1536 | ~3:1 with losses |
| MP3 | floating | up to 48 | up to 320 | 11:1 with losses |
| A.A.C. | floating | up to 96 | up to 529 | with losses |
| Ogg Vorbis | up to 32 | up to 192 | up to 1000 | with losses |
| WMA | up to 24 | up to 96 | up to 768 | 2:1, lossless version available |
Lossless - audio formats with lossless compression, these include:
- FLAC (Free Lossless Audio Codec)
- APE (Monkey's Audio)
- WV (WavPack)
These formats are capable of converting a CD into a digital format while maintaining quality. As an example, you can take a CD, convert it to WAV, then WAV format to FLAC, then back from FLAC to WAV, then burn it to a blank CD and you will have an absolutely identical copy of your source.
In what format does music sound the best?
The most popular is the lossless FLAC format, and one of the most commonly used programs for converting CDs to FLAC format is EAC (Exact Audio Copy).
Of all the parameters of digital audio, you need to pay attention primarily to the following indicators:
sampling frequency (accuracy of digitizing an analog signal over time),
bitrate (the amount of information contained in the file in terms of per second).
The sample rate is the frequency at which digital audio is processed. The most common sampling rate in quality audio formats is 44.1 kHz
It is generally accepted that a high bitrate guarantees best quality- this is true, but only if the source file is of high quality. A high-quality MP3 should have a bitrate of 320 kbps, but a high-quality FLAC format usually has a bitrate of 900 kbps or higher.
What is the best music format in terms of quality?
In addition to the audio formats themselves, for high-quality music sound, you also need high-quality playback equipment: speakers, amplifiers, headphones. In other words, using desktop PC speakers and budget headphones, you will not be able to fully enjoy high-quality sound and unlock the full potential of lossless formats.
Without going deeply into technical details, we can recommend the following formats:
For home listening, I recommend, in my opinion, the best format is FLAC. For Audio Player good decision There will be an MP3 format with a bitrate of at least 320 kbps. Personally, I use only the FLAC format on all devices, fortunately the volumes microSD cards allow you to store a sufficient amount of data in the player.
As for equipment for high-quality music playback, I advise you to pay attention to the following brands:
If budget acoustics do not suit you and you are a fan of high-quality sound (Hi-Fi or Hi-End) equipment, then everything is in your hands and limited only by your budget, I will not give recommendations.
My grandfather listened to the gramophone. My father spent his youth listening to music coming from the speaker of a reel-to-reel tape recorder. My youth saw the rise and fall of cassette recorders. My son is growing up in the era of digital sound. In order to keep up with the times and provide my son with good “sound,” I decided to figure out what determines the quality of playback of a digital audio signal.
I talked to my music lover friends. Conducted an information search on the Internet. As a result, I came to the conclusion that high-quality sound in the digital era can be achieved if you correctly choose the 7 main elements of modern music centers:
- the format in which the music is recorded;
- record player;
- digital-to-analog converter;
- amplifier;
- acoustics;
- cables;
- nutrition.
Below I will share my observations and conclusions regarding achieving high-quality sound from recordings in digital formats.
A lyrical digression, experts don’t have to read it.
I’ll explain in a nutshell where digital sound comes from. During the sound recording process, the microphone converts mechanical vibrations (sound itself) into an analog electrical signal. An analog signal, in the most general case, is similar to the sine wave that we are all familiar with from high school. In the era of analogue sound, it was this signal that was recorded on various media and then reproduced.
With the development of microprocessor technology, it became possible to record and store audio information in digital formats. These formats are obtained using an analog-to-digital conversion (ADC) process.
During the ADC, the analog signal (our sine wave from high school) is converted into a discrete one (in other words, it is cut into parts). At the next stage, the discrete signal is quantized, i.e. each resulting segment of the sinusoid is associated with a digital value. At the third stage, the quantized signal is digitized, i.e. encoded as a sequence of 0 and 1. In relation to digital audio recording, information about the amplitude and frequency of sound is digitized.
Digital audio formats are used to record and store digital audio information. An audio format is a set of requirements for the representation of audio data in digital form.
When discussing sound quality, digital formats are divided into 3 categories:
- Formats without additional compression (CDDA, DSD, WAV, AIFF, etc.);
- Formats compressed without loss of quality (FLAC, WavPack, ADX, etc.);
- Formats that use lossy compression (MP3, AAC, RealAudio, etc.).
High quality sound is obtained when playing music saved in formats from the first and second categories. In the formats of the third category, in order to reduce the volume of data, some information is deliberately excluded. For example, information about hidden frequencies.
Hidden frequencies are those that lie outside the range of perception of the average person: 20 Hz - 22 kHz. For audiophiles, this range, due to individual psychophysiological characteristics, is wider.
To complete your home audio library, you should select recordings saved in files with the extensions:
- *.wav, *.dff, *.dsf, *.aif, *.aiff – these are uncompressed audio files;
- *.mp4, *.flac, *.ape, *.wma are the most common files with lossless compressed audio.
From the history. They say that the very first experiments on sound preservation were carried out by the ancient Greeks. They tried to preserve sound in amphorae. It looked something like this: words were spoken into the amphora and it was quickly sealed. Alas, not one such recording has survived to this day.
When choosing a player, you need to start with an understanding of the form in which your home audio library will be formed. You can buy CDs the old fashioned way or switch to purchasing your favorite music online. The latter option has two significant advantages. It is compact and environmentally friendly:
- The question of space in the apartment for storing CDs does not arise.
- No need to throw faulty disks in the trash.
Have you decided how to buy music? Great! If you buy CDs, you need a CD player. If you prefer online shopping, look for a player on a hard drive or flash memory. Undecided? Great! Look for a universal player. On this you can listen to both discs and files purchased online.
Naturally, you can turn it into a player and a personal computer. But this option is convenient when the computer is truly personal. The prospect of competition for space at the keyboard and possible conflicts will significantly reduce the pleasure of listening to music in good quality.
When choosing a player, pay special attention to the available connectors. The more connector options, the easier it will be to select other elements of the music center.
The player has read a digital sequence from a CD or file. Now comes the most mathematical moment of digital audio reproduction. The digital signal is converted to analog. This math happens in a DAC, or digital-to-analog converter.
The DAC can be built into the player or implemented as a separate unit. If you want to get high quality sound, you need to opt for the second option. The built-in converter is usually inferior in quality to a separate one. The external DAC has its own power supply, the built-in one is powered from a common source with the player. When using an external DAC, its operation is almost unaffected by interference from the player and amplifier.
External DAC according to circuit design solutions is implemented in 4 main versions:
- Pulse width modulator;
- Resampling scheme;
- Weighing type;
- Ladder type, or R-2R chain circuit.
With such a wealth of choice to achieve high quality sound, the R-2R option appears to have no alternative. Due to a special circuit implemented using precision resistances, the ladder-type DAC can achieve very high conversion accuracy.
When choosing an external digital-to-analog converter, you should pay attention to two main characteristics:
- Bit depth. It’s good if the selected model has 24 bits.
- Maximum sampling rate. Very good value 96 kHz, excellent 192 kHz.
To achieve high-quality sound, you need to buy an amplifier along with the speaker system. Essentially, these two elements of the audio center work as one.
A little theory. An amplifier is a device that is designed to increase the power of analog signals audio frequency. It allows you to match the signal received from the DAC with the capabilities of the acoustics. Based on the type of power elements, power amplifiers are divided into tube and transistor ones. Each group contains devices with and without feedback feedback. The introduction of feedback is aimed at correcting distortions that the amplifier itself introduces into the amplified signal. However, when obtaining sound without distortion, you have to accept the loss of some dynamic range sound.
From the point of view of selecting the acoustics-amplifier tandem, it is important to classify the latter according to the type of characteristics of the power element. There are amplifiers with triode and pentode characteristics. Pentode amplifiers come in tube and transistor versions. They are suitable for bookshelf or simple floor-standing speaker systems. For sensitive floor acoustics with a range of 90 dB or more, it is better to select amplifiers with a triode characteristic.
Even before purchasing, you need to try to achieve the ideal balance between the capabilities of the amplifier and acoustics. It’s best to ask the consultants directly in the store to test the selected speaker system together with different amplifiers. You need to choose the set that suits your ear best.
What is a good speaker system is the most confusing question. The choice of acoustics depends on the individual characteristics of a person’s hearing, the parameters of the room in which the system will be placed, and financial capabilities. In this three-variable system, finding a middle ground is very difficult. Therefore, we will consider three fundamental options for solving the problem.
Solution one. Budget. You can equip your home audio center with bookshelf speaker systems. These small systems can be placed on a bookshelf. They are convenient for a small room. Due to its small size, it is also an inexpensive option. A significant disadvantage of this solution is that “shelf” acoustics will not produce normal bass sound.
Solution two. Luxurious. If the dimensions of the room and financial capabilities allow, then you can buy floor-standing acoustics. This system, due to its size, can contain a large diameter woofer. This means there is a chance to enjoy good bass.
Solution three. "Golden" compromise. This solution is suitable for large and small rooms and is affordable. It consists of purchasing a subwoofer and satellites. The subwoofer is responsible for high-quality bass reproduction. Stellites reproduce high frequencies.
When choosing acoustics, you should not follow any advice. You need to rely only on your own hearing. You also need to be prepared for the fact that the sound of the acoustics in the store and in your apartment will be different.
The choice of connecting conductors is an issue that will inevitably have to be resolved to achieve high-quality sound. Many articles have been written about the effect of cables on sound. The only thing the authors achieved unity on was the requirement for cable length. The shorter the better - this is the golden rule when choosing connecting cables.
A little theory. Cables are divided into interconnect and acoustic cables. Interblocks are used to connect audio center blocks, such as a player and a DAC. Speaker cables are used to connect speaker system to the power amplifier.
Based on the type of conductor material, cables are divided into OFC, OCC and composite. OFC are oxygen-free copper cables produced by the pulling method. OCC are cables made from monocrystalline copper obtained directly from the melt. Composite cables are cables in which the conductor consists of several materials.
If you set out to create the perfect audio center from units from different manufacturers, try to use connecting cables that are as short as possible. And be prepared to experiment to achieve the perfect sound quality.
Finally, our home complex for high-quality music playback in digital format is assembled. Now all that remains is a mere trifle. Good equipment requires high-quality power supply. If the most expensive “brand” amplifiers, DACs, players are powered from shared network, then nothing high-quality sound there is no question. Voltage contaminated with interference will kill all efforts to select and purchase high-quality units for the audio center.
Organize power supply for each unit with a separate cable. The cables must be connected directly to the distribution panel at the entrance to the home. Connection sockets must provide high degree fixing the plug. It is wise to use a surge protector; it will make the power supply, and therefore the sound, cleaner.
Target. Understand the process of converting sound information, master the concepts necessary to calculate the volume of sound information. Learn to solve problems on a topic.
Goal-motivation. Preparation for the Unified State Exam.
Lesson Plan
1. View a presentation on the topic with comments from the teacher. Annex 1
Presentation material: Coding audio information.
Since the early 90s personal computers got the opportunity to work with audio information. Every computer that has a sound card, microphone and speakers can record, save and play audio information.
The process of converting sound waves into binary code in computer memory:
The process of reproducing audio information stored in computer memory:
Sound is a sound wave with continuously changing amplitude and frequency. The greater the amplitude, the louder it is for a person; the higher the frequency of the signal, the higher the tone. Computer software now allows a continuous audio signal to be converted into a sequence of electrical pulses that can be represented in binary form. In the process of encoding a continuous audio signal, it is time sampling . A continuous sound wave is divided into separate small temporary sections, and for each such section a certain amplitude value is set.
Thus, the continuous dependence of the signal amplitude on time A(t) is replaced by a discrete sequence of volume levels. On the graph, this looks like replacing a smooth curve with a sequence of “steps”. Each “step” is assigned a sound volume level value, its code (1, 2, 3, etc.
Further). Sound volume levels can be considered as a set of possible states; accordingly, the more volume levels are allocated during the encoding process, the more information the value of each level will carry and the better the sound will be.
Audio adapter ( sound card) is a special device connected to a computer, designed to convert electrical vibrations of sound frequency into a numerical binary code when inputting sound and for inverse conversion(from a numerical code into electrical vibrations) when playing sound.
During the sound recording process, the audio adapter measures the amplitude with a certain period electric current and enters the binary code of the received value into the register. Then the resulting code from the register is rewritten into the computer's RAM. The quality of computer sound is determined by the characteristics of the audio adapter:
- Sampling frequency
- Bit depth (sound depth).
Time sampling rate
This is the number of measurements of the input signal in 1 second. Frequency is measured in Hertz (Hz). One measurement per second corresponds to a frequency of 1 Hz. 1000 measurements in 1 second – 1 kilohertz (kHz). Typical sampling rates of audio adapters:
11 kHz, 22 kHz, 44.1 kHz, etc.
Register width (sound depth) is the number of bits in the audio adapter register that specifies the number of possible sound levels.
The bit depth determines the accuracy of the input signal measurement. The larger the bit depth, the smaller the error of each individual conversion of the electrical signal value into a number and back. If the bit depth is 8 (16), then when measuring the input signal, 2 8 = 256 (2 16 = 65536) different values can be obtained. Obviously, a 16-bit audio adapter encodes and reproduces sound more accurately than an 8-bit one. Modern sound cards provide 16-bit audio encoding depth. The number of different signal levels (states for a given encoding) can be calculated using the formula:
N = 2 I = 2 16 = 65536, where I is the sound depth.
Thus, modern sound cards can provide encoding of 65536 signal levels. Each audio signal amplitude value is assigned a 16-bit code. At binary coding A continuous audio signal is replaced by a sequence of discrete signal levels. The quality of encoding depends on the number of signal level measurements per unit time, that is sampling rates. The more measurements are made in 1 second (the higher the sampling frequency, the more accurate the binary coding procedure.
Sound file - a file that stores audio information in numeric binary form.
2. Repeat the units of measurement of information
1 byte = 8 bits
1 KB = 2 10 bytes = 1024 bytes
1 MB = 2 10 KB = 1024 KB
1 GB = 2 10 MB = 1024 MB
1 TB = 2 10 GB = 1024 GB
1 PB = 2 10 TB = 1024 TB
3. Reinforce the material learned by watching a presentation or textbook
4. Problem solving
Textbook, showing the solution at the presentation.
Task 1. Determine the information volume of a stereo audio file with a sound duration of 1 second with high sound quality (16 bits, 48 kHz).
Task (independently). Textbook, showing the solution at the presentation.
Determine the information volume of a digital audio file with a sound duration of 10 seconds at a sampling frequency of 22.05 kHz and a resolution of 8 bits.
5. Consolidation. Solving problems at home, independently in the next lesson
Determine the amount of memory to store a digital audio file whose playing time is two minutes at a sampling frequency of 44.1 kHz and a resolution of 16 bits.
The user has a memory capacity of 2.6 MB. It is necessary to record a digital audio file with a sound duration of 1 minute. What should the sampling frequency and bit depth be?
The amount of free memory on the disk is 5.25 MB, the bit depth of the sound card is 16. What is the duration of the sound of a digital audio file recorded with a sampling frequency of 22.05 kHz?
One minute of recording a digital audio file takes up 1.3 MB of disk space, and the sound card's bit capacity is 8. At what sampling rate is the sound recorded?
How much memory is required to store a high-quality digital audio file with a playing time of 3 minutes?
The digital audio file contains low-quality audio recording (the sound is dark and muffled). What is the duration of a file if its size is 650 KB?
Two minutes of recording a digital audio file takes up 5.05 MB of disk space. Sampling frequency - 22,050 Hz. What is the bit depth of the audio adapter?
Volume free memory on disk - 0.1 GB, sound card bit depth - 16. What is the duration of the sound of a digital audio file recorded with a sampling frequency of 44,100 Hz?
Answers
No. 92. 124.8 seconds.
No. 93. 22.05 kHz.
No. 94. High sound quality is achieved with a sampling frequency of 44.1 kHz and an audio adapter bit depth of 16. The required memory size is 15.1 MB.
No. 95. The following parameters are typical for a gloomy and muffled sound: sampling frequency - 11 kHz, audio adapter bit depth - 8. The sound duration is 60.5 s.
No. 96. 16 bits.
No. 97. 20.3 minutes.
Literature
1. Textbook: Computer Science, problem book-workshop, volume 1, edited by I.G. Semakin, E.K. Henner)
2. Festival of pedagogical ideas “Open Lesson” Sound. Binary coding of audio information. Supryagina Elena Aleksandrovna, computer science teacher.
3. N. Ugrinovich. Computer science and information technology. 10-11 grades. Moscow. Binomial. Knowledge Laboratory 2003.
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