Goals:
educational:
- Get acquainted with the technology of binary encoding of Wav files
- Learn to solve volumetric problems sound file WAV format
Time sampling - a process in which, during encoding of a continuous sound 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 bit depth, the smaller the error of each individual value conversion electric current to a number and back. 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 the encoded one minute of audio information will occupy when different frequencies sampling:
Algorithm 1 (Calculate the information volume of a sound file):
1) find out how many total values are read into memory during the playing time of the file;
2) find out the code capacity (how many bits in memory each measured value occupies);
3) multiply the results;
4) convert the result into bytes;
5) convert the result to K bytes;
6) convert the result to M bytes;
Algorithm 2 (Calculate the playing time of the file.)
1) Convert the information volume of the file into K bytes.
2) Convert the information volume of the file into bytes.
3) Convert the information volume of the file into bits.
4) Find out how many values were measured (information volume in bits divided by the bit length of the code).
5) Calculate the number of seconds of sound. (Divide the previous result by the sampling frequency.)
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.
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.
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) = 10584000 bytes = 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?
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 = 2726297.6 bytes
D* I =A/T= 2726297.6 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. Volume free memory on disk - 5.25 MB, sound card bit depth - 16. What is the duration of the sound of a digital audio file recorded with a sampling frequency of 22.05 kHz?
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 sound card’s bit capacity is 8. At what sampling rate is the sound recorded?
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 = 1363148.8 bytes
1363148.8 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?
Solution:
Formula for calculating the bit depth: (memory capacity in bytes): (sounding time in seconds): (sampling frequency):
5.1 MB= 5347737.6 bytes
5347737.6 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?
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 = 10737418.24 bytes
10737418.24 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.
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 24,000 = 384,000 bits = 48,000 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?
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 = 15876000 bytes = 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?
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 bits x 48,000 x 2 = 1,536,000 bits = 187.5 KB (multiplied by 2, since 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
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 32,000 = 512,000 bits = 64,000 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)?
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 = 422707200 bytes = 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
256 (2 8) signal intensity levels - radio broadcast sound quality, using 65536 (2 16) 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 frequency - close to the highest 44.1 kHz
Answer: Audio CD quality
b).
1) 157 KB = 160768 bytes = 1286144 bits
2) 1286144 bits: 10 sec = 128614.4 bits/s
3) 128614.4 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) when high quality sound: stereo, 16 bit, 48 kHz.
(, p. 77, No. 2.85)
Solution:
A).
8 bits x 8,000 = 64,000 bits = 8,000 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 bits x 48,000 x 2= 1,536,000 bits = 192,000 bytes = 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. Those. 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 (from 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 of the sound of a symphony orchestra involves more high frequency sampling 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 2 3 = 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 encoding, we get the following result (spaces are included for ease of perception): 100 100 000 011 111 010 011 100 010 110.
Note. It is advisable to draw students' attention to how inaccurately the code conveys the change in amplitude. That is, the sampling rate of 10 Hz and the quantization level of 2 3 (3 bits) 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 2 8 (code 8 bits long).
Answer: 100 100 000 011 111 010 011 100 010 110.
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”; repeat with students binary coding and code length. 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 measuring tool must also support the selected level of quantization (there is no point in representing the length measured with a ruler with decimeter divisions with an accuracy of millimeter).
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.
Quality of the received digital audio 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 more measurements are taken in one second (the higher the sampling frequency), the more accurately the “ladder” of the digital audio signal follows the curve of the analog signal.
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 sound frequencies with great intensity. 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, original form).
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 0s and 1s. 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 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 audio signals. 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 - here it is Golden Rule selection of 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 there can be no talk of any high-quality sound. 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. Use wisely network filter, it will make the power, and therefore the sound, cleaner.
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 a computer to process sound, the continuous audio signal must be converted into 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. The most low quality digitized sound corresponding to the 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 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) with high sound quality: stereo, 16 bits, 48,000 measurements per second.
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 better 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 speakers for PC 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:
I recommend it for home listening in my opinion. best format 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.
Loading...
