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
§ Coefficient nonlinear distortion
§ Intermodulation distortion
§ Uneven amplitude-frequency characteristics
§ Interpenetration of channels
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.)
§ overwrite from one format to another
§ convert convert different audio codecs
Some simple programs, allow only the conversion of formats and codecs.
Varieties of digital audio formats
There are different concepts of audio 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
§ lossless audio formats (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 at the software 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 speech representation
§ 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 software level is a specialized computer program, a codec, that compresses (compresses) or decompresses (decompresses) digital audio data in accordance with 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 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) - a proprietary codec for audio recordings (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 others mobile devices
The human ear perceives sound at frequencies ranging from 20 vibrations per second (low sound) to 20,000 vibrations per second (high sound).
A person can perceive sound in a huge range of intensities, in which the maximum intensity is 10 14 times greater than the minimum (one hundred thousand billion times). A special unit is used to measure sound volume "decibel"(dbl) (Table 5.1). A decrease or increase in sound volume by 10 dbl corresponds to a decrease or increase in sound intensity by 10 times.
Time sampling of sound. In order for the computer to process sound, continuous sound signal must be converted to digital discrete form using time sampling. 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” (Fig. 1.2).
 |
| Rice. 1.2. Time sampling of audio |
Sampling frequency. A microphone connected to the sound card is used to record analog audio and convert it to digital form. The quality of the resulting digital sound depends on the number of measurements of the sound volume level per unit time, i.e. sampling rates. The more measurements are made per second (the higher the sampling frequency), the more accurately the “ladder” of the digital audio signal follows the curve of the dialog signal.
Audio sampling rate is the number of sound volume measurements in one second.
Audio sampling rates can range from 8,000 to 48,000 sound volume measurements per second.
Audio encoding depth. Each “step” is assigned a specific sound volume level. Sound loudness levels can be considered as a set of possible states N, the encoding of which requires a certain amount of information I, which is called the sound 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 formula N = 2 I. Let the audio encoding depth be 16 bits, then 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. The higher the frequency and sampling depth of the sound, the higher the quality of the digitized sound. Lowest quality digitized audio corresponding to quality telephone communication, obtained at a sampling rate of 8000 times per second, a sampling depth of 8 bits and recording one audio track (mono mode). 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 greater the information volume of the sound file. You can 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 in 1 second and multiplied by 2 (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. Digitized sound is presented in sound editors in a visual form, so operations of copying, moving and deleting parts of the audio track can be easily carried out using the mouse. In addition, you can overlay audio tracks on top of each other (mix sounds) and apply various acoustic effects (echo, playback in reverse, etc.).
Sound editors allow you to change the quality of digital audio and the size of the audio file by changing the sampling rate and encoding depth. Digitized audio can be saved uncompressed in audio files in a universal format WAV or in compressed format MP3.
When saving sound in compressed formats, low-intensity sound frequencies that are “excessive” for human perception and coincide in time with high-intensity sound frequencies are discarded. The use of this format allows you to compress sound files tens of times, but leads to irreversible loss of information (files cannot be restored to their original form).
Control questions
1. How do sample rate and encoding depth affect the quality of digital audio?
Tasks for independent completion
1.22. Selective response task. The sound card produces binary encoding of the analog audio signal. How much information is needed to encode each of the 65,536 possible signal intensity levels?
1) 16 bits; 2) 256 bits; 3) 1 bit; 4) 8 bits.
1.23. A task with a detailed answer. Estimate the information volume of digital audio files lasting 10 seconds at a coding depth and audio signal sampling frequency that provide minimum and maximum sound quality:
a) mono, 8 bits, 8000 measurements per second;
b) stereo, 16 bits, 48,000 measurements per second.
1.24. A task with a detailed answer. Determine the duration of the sound file that will fit on a 3.5" floppy disk (note that 2847 sectors of 512 bytes each are allocated for storing data on such a floppy disk):
a) with low sound quality: mono, 8 bits, 8000 measurements per second;
b) when high quality sound: stereo, 16 bits, 48,000 measurements per second.
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 audio frequency into a numerical binary code when inputting sound and for the reverse 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. When binary coding a continuous audio signal, it 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 information volume digital audio a 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, the playing time of which 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?
The amount of free memory on the disk is 0.1 GB, 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 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.
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 the 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 lossy |
| 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 you are not satisfied 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.
Coding of audio information.
Types of tasks:
1. Digital audio file size (mono and stereo).
When solving problems, students rely on the following concepts:
Time sampling – a process in which, during encoding of a continuous audio signal, the sound wave is divided into separate small time sections, and for each such section a certain amplitude value is set. The greater the amplitude of the signal, the louder the sound.
Audio depth (encoding depth) -number of bits per audio encoding.
Volume levels (signal levels)- sound can have different volume levels. The number of different volume levels is calculated using the formula N= 2 I WhereI– depth of sound.
Sampling frequency - number of measurements of the input signal level per unit of time (per 1 second). The higher the sampling rate, the more accurate the binary encoding procedure. Frequency is measured in Hertz (Hz). 1 measurement per 1 second -1 Hz.
1000 measurements in 1 second 1 kHz. Let's denote the sampling rate by the letterD. For encoding, choose one of three frequencies:44.1 KHz, 22.05 KHz, 11.025 KHz.
It is believed that the range of frequencies that a person hears is from 20 Hz to 20 kHz.
Binary encoding quality –a value that is determined by the coding depth and sampling frequency.
Audio adapter (sound card) – a device that converts electrical vibrations of sound frequency into a numerical binary code when inputting sound and vice versa (from a numerical code into electrical vibrations) when playing sound.
Audio adapter specifications:sampling frequency and register bit depth.).
Register size - number of bits in the audio adapter register. The larger the digit capacity, the smaller the error of each individual conversion of the magnitude of the electric current into a number and vice versa. If the bit depth is I, then when measuring the input signal 2 can be obtainedI = N different meanings.
Digital mono audio file size (A) is measured by the formula:
A= D* T* I/8 , WhereD – sampling frequency (Hz),T– time of sound playing or recording,Iregister width (resolution). According to this formula, the size is measured in bytes.
Digital stereo audio file size (A) is measured by the formula:
A=2* D* T* I/8 , the signal is recorded for two speakers, since the left and right sound channels are encoded separately.
It is useful for students to give Table 1, showing how many MB an encoded one minute of audio information will occupy at different sampling rates:
Signal type | Sampling frequency, kHz |
||
16 bit, stereo | |||
16 bit, mono | |||
8 bit, mono |
1. Digital file size
Level "3"
1. Determine the size (in bytes) of a digital audio file whose playing time is 10 seconds at a sampling rate of 22.05 kHz and a resolution of 8 bits. The file is not compressed. (, page 156, example 1)
Solution:
Formula for calculating size (in bytes) digital audio file: A= D* T* I/8.
To convert to bytes, the resulting value must be divided by 8 bits.
22.05 kHz =22.05 * 1000 Hz =22050 Hz
A= D* T* I/8 = 22050 x 10 x 8 / 8 = 220500 bytes.
Answer: The file size is 220500 bytes.
2. Determine the amount of memory to store a digital audio file, the playing time of which is two minutes at a sampling frequency of 44.1 kHz and a resolution of 16 bits. (, p. 157, no. 88)
Solution:
A= D* T* I/8. – the amount of memory for storing a digital audio file.
44100 (Hz) x 120 (s) x 16 (bits) / 8 (bits) = byte = 10335.9375 KB = 10.094 MB.
Answer: ≈ 10 MB
Level "4"
3. 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? (, p. 157, no. 89)
Solution:
Formula for calculating the sampling frequency and bit depth: D* I =A/T
(memory capacity in bytes) : (sounding time in seconds):
2.6 MB = 26 bytes
D* I =A/T= 26 bytes: 60 = 45438.3 bytes
D=45438.3 bytes: I
The adapter width can be 8 or 16 bits. (1 byte or 2 bytes). Therefore the sampling frequency can be either 45438.3 Hz = 45.4 kHz ≈ 44.1 kHz–standard characteristic sampling frequency, or 22719.15 Hz = 22.7 kHz ≈ 22.05 kHz- standard characteristic sampling rate
Answer:
Sampling frequency | Audio adapter capacity |
|
1 option | 22.05 kHz | 16 bit |
Option 2 | 44.1 kHz | 8 bit |
4. 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? (, p. 157, no. 90)
Solution:
Formula for calculating sound duration: T=A/D/I
(memory capacity in bytes) : (sampling frequency in Hz) : (sound card capacity in bytes):
5.25 MB = 5505024 bytes
5505024 bytes: 22050 Hz: 2 bytes = 124.8 sec
Answer: 124.8 seconds
5. One minute of recording a digital audio file takes up 1.3 MB of disk space, the bit depth of the sound card is 8. At what sampling rate is the sound recorded? (, p. 157, no. 91)
Solution:
Formula for calculating the sampling rate: D = A/T/I
(memory capacity in bytes) : (recording time in seconds) : (sound card capacity in bytes)
1.3 MB = 18 bytes
18 bytes: 60:1 = 22719.1 Hz
Answer: 22.05 kHz
6. Two minutes of recording a digital audio file takes up 5.1 MB of disk space. Sampling frequency - 22050 Hz. What is the bit depth of the audio adapter? (, p. 157, no. 94)
Solution:
Formula for calculating the bit depth: (memory capacity in bytes): (sounding time in seconds): (sampling frequency):
5.1 MB= 56 bytes
56 bytes: 120 sec: 22050 Hz= 2.02 bytes = 16 bits
Answer: 16 bits
7. The amount of free memory on the disk is 0.01 GB, 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 44100 Hz? (, p. 157, no. 95)
Solution:
Formula for calculating sound duration T=A/D/I
(memory capacity in bytes) : (sampling frequency in Hz) : (sound card capacity in bytes)
0.01 GB = .24 bytes
24 bytes: 44100: 2 = 121.74 sec = 2.03 min
Answer: 20.3 minutes
8. Estimate the information volume of a mono audio file with a sound duration of 1 minute. if the encoding “depth” and the audio signal sampling frequency are equal, respectively:
a) 16 bits and 8 kHz;
b) 16 bits and 24 kHz.
(, p. 76, No. 2.82)
Solution:
A).
16 bits x 8,000 = 128,000 bits = 16,000 bytes = 15.625 KB/s
15.625 KB/s x 60 s = 937.5 KB
b).
1) The information volume of a sound file lasting 1 second is equal to:
16 bits x = 384000 bits = 48000 bytes = 46.875 KB/s
2) The information volume of a sound file lasting 1 minute is equal to:
46.875 KB/s x 60 s = 2812.5 KB = 2.8 MB
Answer: a) 937.5 KB; b) 2.8 MB
Level "5"
Table 1 is used
9. How much memory is required to store a digital audio file with high-quality sound recording, provided that the playing time is 3 minutes? (, p. 157, no. 92)
Solution:
High sound quality is achieved at a sampling frequency of 44.1 kHz and an audio adapter bit depth of 16.
Formula for calculating memory capacity: (recording time in seconds) x (sound card capacity in bytes) x (sampling frequency):
180 s x 2 x 44100 Hz = byte = 15.1 MB
Answer: 15.1 MB
10. 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? (, p. 157, no. 93)
Solution:
The following parameters are typical for gloomy and muffled sound: sampling frequency - 11.025 KHz, audio adapter bit depth - 8 bits (see Table 1). Then T=A/D/I. Let's convert the volume into bytes: 650 KB = 665600 bytes
Т=665600 bytes/11025 Hz/1 byte ≈60.4 s
Answer: the duration of the sound is 60.5 s
Solution:
The information volume of a sound file lasting 1 second is equal to:
16 bit xx 2 = 1 bit = 187.5 KB (multiplied by 2, since it is stereo).
The information volume of a sound file lasting 1 minute is equal to:
187.5 KB/s x 60 s ≈ 11 MB
Answer: 11 MB
Answer: a) 940 KB; b) 2.8 MB.
12. Calculate the playing time of a mono audio file if, with 16-bit encoding and a sampling frequency of 32 kHz, its volume is equal to:
a) 700 KB;
b) 6300 KB
(, p. 76, No. 2.84)
Solution:
A).
1) The information volume of a sound file lasting 1 second is equal to:
700 KB: 62.5 KB/s = 11.2 s
b).
1) The information volume of a sound file lasting 1 second is equal to:
16 bits x = 512000 bits = 64000 bytes = 62.5 KB/s
2) The playing time of a 700 KB mono audio file is:
6300 KB: 62.5 KB/s = 100.8 s = 1.68 min
Answer: a) 10 seconds; b) 1.5 min.
13. Calculate how many bytes of information one second of stereo recording occupies on a CD (frequency 44032 Hz, 16 bits per value). How long does one minute take? What is the maximum disk capacity (assuming a maximum duration of 80 minutes)? (, p. 34, exercise No. 34)
Solution:
Formula for calculating memory size A=
D*
T*
I:
(recording time in seconds) * (sound card capacity in bytes) * (sampling frequency). 16 bits -2 bytes.
1) 1s x 2 x 44032 Hz = 88064 bytes (1 second stereo CD recording)
2) 60s x 2 x 44032 Hz = 5283840 bytes (1 minute of stereo CD recording)
3) 4800s x 2 x 44032 Hz = byte = 412800 KB = 403.125 MB (80 minutes)
Answer: 88064 bytes (1 second), 5283840 bytes (1 minute), 403.125 MB (80 minutes)
2. Determination of sound quality.
To determine the sound quality, you need to find the sampling frequency and use table No. 1
signal intensity levels - radio broadcast sound quality, using 65 signal intensity levels - audio CD sound quality. The highest quality frequency corresponds to music recorded on a CD. The magnitude of the analog signal is measured in this case 44,100 times per second.
Level "5"
13. Determine the sound quality (radio broadcast quality, average quality, audio CD quality) if it is known that the volume of a mono audio file with a sound duration of 10 seconds. equal to:
a) 940 KB;
b) 157 KB.
(, p. 76, No. 2.83)
Solution:
A).
1) 940 KB = 962560 bytes = 7700480 bits
2) 7700480 bits: 10 sec = 770048 bits/s
3) 770048 bps: 16 bits = 48128 Hz – sampling rate – close to the highest 44.1 kHz
Answer: Audio CD quality
b).
1) 157 KB = 160768 bytes = 1286144 bits
2) 1286144 bits: 10 sec = 4 bps
3) 4 bps: 16 bits = 8038.4 Hz
Answer: broadcast quality
Answer: a) CD quality; b) quality of radio broadcast.
14. Determine the length of the audio file that will fit on a 3.5” floppy disk. Please note that 2847 sectors of 512 bytes are allocated to store data on such a floppy disk.
a) with low sound quality: mono, 8 bit, 8 kHz;
b) with high sound quality: stereo, 16 bit, 48 kHz.
(, p. 77, No. 2.85)
Solution:
A).
8 bits x 8,000 = bits = 8000 bytes = 7.8 KB/s
3) The playing time of a mono audio file with a volume of 1423.5 KB is equal to:
1423.5 KB: 7.8 KB/s = 182.5 s ≈ 3 min
b).
1) The information volume of a floppy disk is equal to:
2847 sectors x 512 bytes = 1457664 bytes = 1423.5 KB
2) The information volume of a sound file lasting 1 second is equal to:
16 bit xx 2= 1 bit = byte = 187.5 KB/s
3) The playing time of a stereo audio file with a volume of 1423.5 KB is equal to:
1423.5 KB: 187.5 KB/s = 7.6 s
Answer: a) 3 minutes; b) 7.6 seconds.
3. Binary audio coding.
When solving problems, he uses the following theoretical material:
In order to encode audio, the analog signal shown in the figure
the plane is divided into vertical and horizontal lines. Vertical partitioning is the sampling of the analog signal (signal measurement frequency), horizontal partitioning is quantization by level. That is, the finer the grid, the better the approximation of analog sound using numbers. Eight-bit quantization is used to digitize ordinary speech ( telephone conversation) and radio broadcasts on short waves. Sixteen-bit – for digitizing music and VHF (ultra-short wave) radio broadcasts.
Level "3"
15. The analog audio signal was sampled first using 256 signal intensities (broadcast sound quality) and then using 65,536 signal intensities (audio CD sound quality). How many times do the information volumes of digitized sound differ? (, p. 77, No. 2.86)
Solution:
The code length of an analog signal using 256 signal intensity levels is 8 bits, and using 65536 signal intensity levels is 16 bits. Since the code length of one signal has doubled, the information volumes of the digitized sound differ by a factor of 2.
Answer: 2 times.
Level "4"
16. According to the Nyquist-Kotelnikov theorem, in order for an analog signal to be accurately reconstructed from its discrete representation(according to its samples), the sampling frequency must be at least twice the maximum audio frequency of this signal.
· What should be the sampling rate of human-perceivable sound?
· Which should be higher: the sampling rate of speech or the sampling rate of a symphony orchestra?
Goal: To introduce students to the characteristics of hardware and software for working with sound. Types of activities: attracting knowledge from a physics course (or working with reference books). (, p. ??, task 2)
Solution:
It is believed that the range of frequencies that humans hear is from 20 Hz to 20 kHz. Thus, according to the Nyquist-Kotelnikov theorem, in order for an analog signal to be accurately reconstructed from its discrete representation (from its samples), The sampling rate must be at least twice the maximum audio frequency of that signal. The maximum sound frequency that a person can hear is 20 KHz, which means that the device ra and software must provide a sampling rate of at least 40 kHz, or more precisely 44.1 kHz. computer processing The sound of a symphony orchestra requires a higher sampling rate than speech processing, since the frequency range in the case of a symphony orchestra is much larger.
Answer: no less than 40 kHz, the sampling frequency of a symphony orchestra is higher.
Level "5"
17. The figure shows the sound of 1 second of speech recorded by a recorder. Encode it in binary digital code with a frequency of 10 Hz and a code length of 3 bits. (, p. ??, task 1)
Solution:
Encoding at 10 Hz means we have to measure the pitch 10 times per second. Let's choose equidistant moments of time:
A code length of 3 bits means 23 = 8 quantization levels. That is, as a numerical code for the pitch of the sound at each selected moment in time, we can set one of the following combinations: 000, 001, 010, 011, 100, 101, 110, 111. There are only 8 of them, therefore, the pitch of the sound can be measured at 8 " levels":
We will “round” the pitch values to the nearest lower level:
Using this method coding, we get the following result (spaces are included for ease of perception):
Note. It is advisable to draw students' attention to how inaccurately the code conveys the change in amplitude. That is, the sampling frequency of 10 Hz and the bit quantization level) are too small. Typically, for sound (voice), a sampling frequency of 8 kHz is chosen, i.e. 8000 times per second, and a quantization level of 28 (8-bit code).
Answer:
18. Explain why the quantization level is, along with the sampling frequency, the main characteristics of sound representation in a computer. Goals: to consolidate students’ understanding of the concepts of “accuracy of data representation”, “measurement error”, “representation error”; Review binary coding and code length with students. Type of activity: working with definitions of concepts. (, p. ??, task 3)
Solution:
In geometry, physics, and technology, there is the concept of “measurement accuracy,” which is closely related to the concept of “measurement error.” But there is also a concept "precision of representation". For example, about a person’s height we can say that he is: a) about. 2 m, b) slightly more than 1.7 m, c) equal to 1 m 72 cm, d) equal to 1 m 71 cm 8 mm. That is, 1, 2, 3 or 4 digits can be used to indicate measured height.
The same goes for binary encoding. If only 2 bits are used to record the pitch of a sound at a particular moment in time, then, even if the measurements were accurate, only 4 levels can be transmitted: low (00), below average (01), above average (10), high (11). If you use 1 byte, you can transfer 256 levels. How higher quantization level, or, which is the same as The more bits allocated to record the measured value, the more accurately this value is transmitted.
Note. It should be noted that the measuring instrument must also support the selected quantization level (there is no point in representing the length measured with a ruler with decimeter divisions with an accuracy of a millimeter).
Answer: the higher the quantization level, the more accurately the sound is transmitted.
Literature:
[ 1] Computer science. Problem book-workshop in 2 volumes /Ed. , : Volume 1. – Laboratory Basic Knowledge, 1999 – 304 p.: ill.
Workshop on computer science and information technology. Tutorial for educational institutions / , . – M.: Binom. Laboratory of Knowledge, 20 p.: ill.
Informatics at school: Supplement to the journal “Informatics and Education”. No. 4 - 2003. - M.: Education and Informatics, 2003. - 96 p.: ill.
Etc. Information culture: information coding. Information models. Grades 9-10: Textbook for general education institutions. - 2nd ed. - M.: Bustard, 1996. - 208 p.: ill.
Senokosov on computer science for schoolchildren. - Ekaterinburg: “U-Factoria”, 2003. - 346. p54-56.
