Tag: what do analog and digital have in common

Analog-Digital Processing


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Analog-Digital Processing

Digital vs Analog

A digital signal is obtained from analog or is directly synthesized into digital (in electric musical instruments).

DIGITAL ANALOG AUDIO

Converting from analog to digital involves two basic operations: sampling and quantizing. Discretization is the replacement of a continuous signal with a series of samples of its instantaneous values ​​taken at regular intervals. According to the Kotelnikov-Chenon theorem, a discrete signal can be completely restored later, as long as the sampling frequency is at least twice the upper frequency of the signal spectrum. The samples are then quantized according to level: each of them is assigned a discrete value closer to the real one. The precision of quantization is determined by the bit width of the binary representation. The higher the bit depth, the more quantization levels (2N,

The audio CD format has a sampling frequency of 44.1 kHz and 16 bits. This gives 44 thousand samples per second, each of which can take one of 216 = 65536 levels (for each of the stereo channels).

In addition to the 44.1 kHz / 16-bit format, others are used in digital recording. Studio recording is generally done in 20-24 bit, then the data is converted to audio CD by recalculation. The extra bits are then discarded or (better) rounded, sometimes pseudo-random noise is added to reduce quantization noise (dither).

The most advanced custom audio formats are DVD Audio and Super Audio CD (SACD). DVD Audio adopts the MLP lossless data compression algorithm developed by Meridian. And SACD, unlike other formats, does not use pulse code modulation (PCM or PCM), but one-bit encoding of the DSD (Discrete Pulse Width Modulation) stream. SACDs come in single or double layer (hybrid) discs with a normal CD layer.

The most popular audio medium today is compact disc, despite certain limitations in sound quality seen by audiophiles. The reason for them is in the low sample rate: for an accurate reconstruction of signals near the upper limit of the audio range, a filter that is not physically workable is needed (its impulse response covers the negative time area). This is compensated to some extent by digital filtering with higher sampling and bit depth. The data on the disc is redundantly encoded (Reed-Solomon code) to ensure smooth playback in real time.

Broadband communication is required for digital audio transmission, especially for uncompressed high definition multichannel transmissions.

Figure: 1. Digitizing an analog signal and obtaining digital samples on CD Audio and SACD (right)

DIGITAL AUDIO TRANSMISSION

The communication lines for digital audio transmission can be cables, optical lines, and overhead radio.

For the transmission of PCM signals over wired lines, AES / EBU (balanced, coaxial), S / PDIF (unbalanced coaxial) interfaces have been developed, which provide transmission of various signals (clock frequency, digital word rate, channel data) over a cable. Inside the devices, these signals are transmitted separately, at the output of the transport mechanism they are encoded and at the input of a digital-to-analog converter (in two-block systems) they are separated again in a digital receiver.

Typically, a high-quality coaxial cable is used for digital audio transmission. There are also S / PDIF converters for fiber optic lines: AT&T ST and Toslink (the latter is standard in consumer equipment). And also, for the use of twisted pairs in Ethernet cable networks. The medium of distribution for compressed audio in the form of archived files is the Internet.

Like any digital signal, digitized audio is distributed and switched by special devices: distribution amplifiers, matrix switches, and conventional.

There is one factor that negatively affects digital signals, and often negates almost all of the advantages of digital audio over analog, including the ability to repeatedly copy, stream, and archive programs without any loss of quality: we are talking about jitter. Jitter is jitter, or the uncertainty of a transition from 0 to 1 and vice versa.


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Working with sound

Working with sound

Analog and Digital

Analog and digital audio

Analog and Digital

Analog sound recording is based on the conversion of acoustic waves into electrical waves using a microphone. A microphone consists of a small membrane that can vibrate and a mechanism to convert the vibrations of the membrane into an electrical signal. (The exact electrical mechanism differs depending on the type of microphone.) Generally, a higher pressure corresponds to a higher voltage and vice versa.

The recorder transmits the waveform one more time, this time from an electrical signal through a wire to a magnetic signal on tape. When the recording is played back, the opposite process occurs: the magnetic signal is converted into an electronic signal, which makes the speaker vibrate (usually electromagnetic).

The main device for digital recording is an analog-to-digital converter (ADC, analog-to-digital converter, ADC). The ADC captures a chunk of electrical voltage on the audio path and presents it as a number, which is then transmitted to the computer. By capturing the voltage several thousand times per second, you can get a signal quite close to the original. The unit of capture is called a sample (each number in a sound file represents corresponds to a sample in a waveform).

There are two factors that determine the quality of a digital recording:

Sampling rate
The frequency at which samples are captured or played, measured in Hertz (Hz) or samples per second. A typical audio CD is recorded at a sample rate of 44100 Hz, more commonly known as 44 kHz for short. This is the same default sample rate used for most digitals.

Sample format (size)
The number of digits in the digital representation of each sample. Imagine that the sample rate is plotted horizontally and the sample size is plotted vertically. Audio CD is 16 bits wide, which corresponds to approximately 5 decimal places.

Higher sample rates for digital recording provide accurate recording at higher frequencies. The sample rate must be at least twice the highest desired sample rate. The average human ear is believed to be unable to distinguish frequencies above 20,000 Hz, so 44,100 Hz was chosen as the standard for audio CDs. Now the transition to the frequencies of 96 and 192 kHz is taking place gradually, in particular within the DVD-Audio format. However, many people just don’t hear the difference between 44.1 kHz and 192 kHz audio.

Larger sample sizes provide a greater dynamic range, that is, the ability to present louder and quieter sounds. If you are familiar with the decibel (dB) scale, you can give an example from ordinary audio CDs – its dynamic range is theoretically 90 dB, but it actually sounds lower than -24 dB. Audacity supports two more sample sizes: 24-bit, which is most often used in digital studio recording, and 32-bit floating point, whose dynamic range covers all imaginable needs, despite the fact that the data with these parameters occupies just twice the disk space compared to 16-bit audio.

When playing digital sound, a digital-to-analog converter (DAC) is used. In this case, to recreate the original signal and then digitized with the ADC, a sample is taken, from which a certain voltage is established at the analog outputs. The first CD players did just that, so the sound quality was not very good. Modern players also smooth out the audio signal by sampling within a range of the sampling frequency. The quality of the filters on the DAC also affects the sound signal that is recreated. The filter is one of the signal adaptation stages in the DAC.

The inevitable loss in the transition from analog to digital audio can be offset by a number of advantages of digital recording. Digital data can be copied as much as you like and there is no loss of quality. This data can be burned to a music CD or posted on the Internet as compressed files. Also, digital recordings are much easier to edit than analog tapes.

A personal computer has all the necessary devices to convert audio data from analog to digital and vice versa. First of all, it’s a sound card, an additionally installed separate device like Creative SBLive !, and maybe a sound chip built into the motherboard. In both cases, the audio device contains an analog-to-digital converter (ADC) to record sound and a digital-to-analog converter (DAC) to play it back. The operating system you are using interacts with the sound card,

Analog and digital sound sources

Analog and digital sound sources

analog and digital audio

Digital music comes from two main sources: analog and digital.

Analog and Digital

Analogous sources
An analog music source must use an analog-to-digital converter, such as a sound card, to convert the physical changes of the analog medium into a digital file that can be read by a computer. An analog medium is an object that stores music in itself through physical changes.

For example:

A cassette recorder changes the degree of magnetization of a cassette tape to record sound. Connecting a cassette deck to a recording device allows you to make a digital copy of an analog cassette.
The recorder cuts grooves in the vinyl record to create a physical representation of the sound. Ripping vinyl with a preamp and sound card allows you to make a digital copy of an analog record.
Analog recordings can be converted to digital music files in various formats, such as FLAC and MP3. Vinyl recordings can always be posted to the site, but posting tape recordings and other analog sources requires approval from the moderator.

Digital music sources
Music from digital sources is already encoded in a computer-compatible format, so no additional conversion is required. A digital medium is an object that stores music digitally (as a sequence of binary numbers).

For example:

CD
DVD
Super Audio CD (SACD)
Content from online stores (iTunes, Amazon, etc.)
Music from digital sources can be uploaded to RED after analyzing the spectrograms of the files to verify lossy transcoding.

Comparison of analog to digital music sources
Controversy still exists as to whether music sounds differently from analog and digital sources. Some people prefer the feel of vinyl and find that music on vinyl sounds “warmer” and “brighter.” On the other hand, some believe that digital sources provide true, pure and authentic sound. Both are represented in RED, so you can compare and make your own choice!

Analog and digital

First of all, a fundamental distinction is necessary: ​​what is meant by an analog signal and what is meant by a digital signal. Sampling is in fact an analog-to-digital conversion, and to understand how this is accomplished, it is necessary to understand what the subjects of this transformation are.

The classic definition of “analog” and “digital” is as follows.
The analog signal is one in which the variation is continuous in time.
The digital signal is one in which the variation in time occurs in a discrete way.
Pay attention to this definition because it expresses a very simple concept but at the same time misunderstood.

Let’s use some examples to get the concept down.
As a first example, let’s think of a watch with hands (suppose it is of the type in which the second hand moves continuously and not broken).

This clock not only marks the hours, minutes and seconds, but also any other type of fraction that we want to imagine: half seconds, tenths, hundredths, etc. As difficult as it is for the eye to distinguish the different moments, we know that the clock continuously passes through every instant of time that we can imagine.

Let’s think instead of a digital clock, those that indicate the time with numbers on a screen. This clock will mark the hours, minutes and seconds, activating the latter one by one; We do not see half seconds, tenths and so on: from 10:10:01 to 10:10:02 (for example) the clock will always read 10:10:01.

The watch with hands can be defined as an analog device, while the other watch, which provides only discrete, but not continuous measurements, is called digital.

A second example: let’s think about two different ways to monitor the level of a signal: the first, the classic needle VU-meter, typical of old mixers; the second, the column of bright LEDs, typical for example of equalizers.

The VU-meter, for reasons exactly analogous to those of the hand watch, is an analog device; The LED column, which only provides discrete data, is a digital device.

So what does it mean to sample a signal?

It means finding a discrete representation for something that originally has continuous variation.
The purpose is obvious: where, for example, to modify the analog recording of a voice, we must first convert the sound energy into electrical energy (through a microphone), then transform the electrical energy into the magnetic property of a tape ( through a tape recorder) and finally intervene with mechanical modifications to the tape itself (editing operations with manual cutting and pasting of the tape), with a digital recording, in which the electrical energy supplied by the microphone is converted directly In digital samples, that is, in discrete number data, it will be possible to modify the register through an electronic calculator capable of analyzing and modifying the data.

Sampling and time (frequency and Nyquist theorem)

The first practical problem that sampling is faced with is establishing how many times in a given period of time the signal must be measured for the sampling to be accurate, and the resulting digital signal can be converted back into an analog signal without losing or changing certain characteristics of the original signal.

Take as an example the classic elementary sinusoid, like the one in the figure.

Let’s say we have a device that takes, over a certain period of time, a certain number of samples of the signal: for example, 14 samples per period of the sinusoid.
We will obtain a series of samples like the one in the figure:

We see that the original sinusoid is still intuitive, so it is possible to reconstruct it and reverse the procedure.
But imagine halving the sample rate, that is, doubling the time between one measurement and another.
We will obtain a different series of samples, less dense than the previous one:

The sine wave can still be guessed, but it is clear that we have lost some of the original information.
Halving again, the situation becomes almost critical:

Here it is already very difficult to trace the original signal.
By reducing more by half, all traces of the sine wave are lost:

Therefore, we understood that there is a critical point, below which the sampling frequency cannot fall, under penalty of total loss of information.