Is the digital signal distorted during transmission and storage?


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Is the digital signal distorted during transmission and storage?

DIGITAL AUDIO

Since any digital signal is represented as a real voltage or current electrical curve, its shape is distorted in one way or another during any transmission, and a signal “frozen” for storage (signalogram) is subject to degradation due to physical reasons. common.

Digital Audio

All of these influences on the shape of the carrier signal are interferences that, up to a certain value, do not change the information content of the signal, since individual distortions and letter loss in words generally do not interfere with the correct understanding of words. words, and information redundancy, such as an increase in the length of the words, increases the probability of successful recognition. … In other words, the carrier signal itself can be distorted, but the information it carries, the encoded audio signal, remains unchanged in the vast majority of cases.

So that the quality of the carrier signal does not deteriorate, any transmission of useful audio information (copying, writing to a carrier and reading it) must necessarily include the operation of restoring the form of the carrier signal, and ideally, and the digital form primary of the information signal, and only after that the newly generated carrier signal can be transmitted to the next consumer. In the case of direct copy without restoration (for example, simply rewriting a video cassette with a digital signal obtained with a PCM decoder in common VCRs), the quality of the digital signal deteriorates, although it still contains all the information it carries. However, after repeated sequential copies or long-term storage, the quality deteriorates so much that unrecoverable errors begin to appear that irreversibly distort the information carried by the signal. Therefore, the copying and transmission of digital signals should be done only on digital devices and, when stored on media, should be “updated” in a timely manner without waiting for irreversible degradation (for magnetic media, this period is estimated to be several years ). A correctly transmitted or updated digital signallogram does not lose quality and can be copied and exist forever in absolutely unaltered form. without waiting for irreversible degradation (for magnetic carriers this period is estimated to be several years). A correctly transmitted or updated digital signallogram does not lose quality and can be copied and exist forever in absolutely unaltered form. without waiting for irreversible degradation (for magnetic carriers this period is estimated to be several years). A correctly transmitted or updated digital signallogram does not lose quality and can be copied and exist forever in absolutely unaltered form.

However, it should not be forgotten that the correctness of any code is finite, and the actual carriers are far from ideal, therefore the occurrence of unrecoverable errors is such a rare thing, especially with careless handling of the carrier. When reading new and correctly stored DAT cassettes or CDs on high-quality and reliable devices, these errors practically do not occur, however, with aging, contamination and damage of media and reading systems, they become more. A single uncorrected error is almost always invisible to the ear due to interpolation, however, it leads to distortion of the original sound signal, and the accumulation of such errors over time begins to be felt in the ear.

A separate problem is the difficulty of recording uncorrected errors, as well as verifying the identity of the original and the copy. Very often, designers of digital audio devices operating in real time do not care about the issue of accurate verification of the reliability of the transmission, considering that the measures taken to correct the errors are sufficient. In the general case, the impossibility of retransmitting an erroneous sample or block leads to interpolation occurring secretly and after copying it is impossible to say with certainty whether the original signal was copied exactly. Error indicators, which are found on some devices, usually light up only at the moment of their appearance, and in the case of single errors, their operation can easily go unnoticed. Even in personal computer-based systems, it is often impossible to control the accuracy of reception through a digital interface or direct reading from a CD; the only way out is to repeat the operation and compare the results.


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What are the pros and cons of digital audio?

What are the pros and cons of digital audio?

Digital Audio

The digital representation of sound is valuable, first of all, for the possibility of endless storage and reproduction without loss of quality; however, the conversion from analog to digital and vice versa inevitably leads to its partial loss.

digital audio

The most unpleasant distortions introduced in the digitizing stage are the granular noise that occurs when the signal is quantized by level due to rounding of the amplitude to the nearest discrete value. Unlike simple broadband noise introduced by quantization errors, granular noise is the harmonic distortion of the signal, most noticeable in the upper part of the spectrum.

The power of the granular noise is inversely proportional to the number of quantization steps; However, due to the logarithmic characteristic of hearing with linear quantization (constant step value), quiet sounds have fewer quantization steps than loud sounds, and as a result, the main density of non-linear distortions falls in the region of sounds. silent. This leads to a limitation of the dynamic range, which ideally (without taking into account harmonic distortion) would be equal to the signal-to-noise ratio, but the need to limit this distortion reduces the dynamic range for 16-bit encoding to 50-60 dB. The situation could have been saved by logarithmic quantification, but its implementation in real time is very difficult and expensive.

The distortion introduced by granular noise can be reduced by adding normal white noise (random or pseudo-random signal) to the signal, with an amplitude of half the least significant bit; such an operation is called dithering. This leads to a slight increase in the noise level, but weakens the correlation of quantization errors with the components of the high-frequency signal and improves subjective perception. Anti-aliasing is also applied before rounding the samples by decreasing their bit depth. Essentially, dithering and noise shaping are special cases of the same technology, with the difference that, in the first case, white noise with a flat spectrum is used and, in the second, noise with a spectrum with a “shape “special.

When restoring audio from digital to analog, there is the problem of smoothing the stepped waveform and suppressing the harmonics introduced by the sample rate. Due to the imperfection of the frequency response of the filters, insufficient suppression of this interference or excessive attenuation of useful high-frequency components may occur. Poorly suppressed sample rate harmonics distort the shape of the analog signal (especially in the high frequency region), resulting in a “rough” and “dirty” sound.

Basics of digital audio

Basics of digital audio:

Before the computer can record, manipulate, and reproduce sound, sound must be transformed from an audible analog form to a computer-acceptable digital form, using a process called analog-to-digital conversion (ADC). Once the sound data has been stored as bytes in the computer, the power of the computer’s CPU can be used to transform this sound in thousands of ways. Finally, when you are ready to listen to the result, the digital-to-analog conversion (DAC) process transforms the sound bytes back into an analog electrical signal from the speakers.

Sampling: Analog to Digital Conversion

Given an analog signal, discrete values ​​of its amplitude are taken at small time intervals, obviously the more reliable the reproduction the more samples per second are taken. These obtained values ​​are assigned a digital value that the computer can understand and process as required. We can use 8 or 16 bit words, thus obtaining 256 or 65536 different combinations and obtaining higher resolution.

 

SAMPLE FREQUENCY: According to the Nyquist theorem, it is possible to accurately repeat a waveform if the sampling frequency is at least twice the frequency of the component with the highest frequency. The highest frequency that the human ear can perceive is close to 20 kHz, so the 44.1 kHz sampling rate of sound cards is more than enough. This value is the one used today by CD audio players.

SAMPLE SIZE: The sample size controls the dynamic range that can be recorded. For example, 8-bit samples limit the dynamic range to 256 steps (50 dB range). In contrast, a 16-bit sample has a dynamic range of 65,536 steps (90 dB range) a substantial improvement. The human ear perceives a whole world of differences between these two sample sizes. Ears are more sensitive to detecting differences in pitch than intensity, but are even more sensitive to the strength of sound.

From the previous processes we can get an audio file, such as (and since it is the best known), a WAV audio file. It is the own format of Windows. They can be 8 or 16 bit with sampling rates of 11,025 kHz, 22.05 kHz, or 44.1 kHz and generally have good sound quality.

Digital audio compression

It could be assumed that all you have to do to get good sound is to record at the 44.1 kHz speed limit with 16-bit (2-byte) samples. The only problem that appears if recording in stereo, sampling simultaneously on the left and right channels at 44.1 kHz, a one minute sound sample needs a 10.58MB storage space. This involves using large disk spaces to store these sound files. Many compressed file formats (codecs) have been developed that enable high-quality recording without the need for so much disk space.

Most common audio formats:

With the simple objective of listing a series of codecs used by different operating systems to perform audio compression. Later, a more complete description of the most used is made: MP3.

Therefore, some of the most used are:

Advanced Audio Coding (AAC): used by Apple computers. More efficient than MP3.

Audio for Unix (AU): Acoustic standard for the JAVA programming language.

Windows Media Audio (WMA)

Ogg Vorbis: It is free, open and not patented.

Atrac: compression and playback technology for minidisc.

 

The codec par excellence: the MP3

Its origin and current

The abbreviations MP3 respond to the abbreviation of MPEG (Moving Picture Expert Group) 1 Layer 3, which is a perceptual coding algorithm. This among others was developed by the Moving Picture Expert Group (MPEG) (http://www.cselt.it/mpeg/) together with the Fraunhofer Institute of Technology (http://www.ipa.fhg.de/english/ ).

Moving Picture Expert Group is an ISO / IEC research committee. MPEG is in charge of the international development of compression, decompression, processing and encoded rendering standards for movies, audio and the combination of both. It is a non-profit institution created in 1988, which brings together 300 experts from 20 countries three times a year.

Introduction to digital audio

Introduction to digital audio

Digital audio is the representation of sound signals through a set
of binary data. A complete digital audio system usually begins
with a transceiver (microphone) that converts the pressure wave that represents the
Sound to an analog electrical signal.
This analog signal goes through an analog signal processing system, in
which can be made limitations on frequency, equalization, amplification and
Other processes such as compassion. Equalization aims
counteract the particular frequency response of the transceiver used of
so that the analog signal closely resembles the original audio signal.


After analog processing, the signal is sampled, quantified and encoded. The
sampling takes a discrete number of analog signal values ​​per second
(sampling rate) and quantification assigns discrete analog values ​​to those
samples, which means a loss of information (the signal is no longer the same
than the original). The encoding assigns a sequence of bits to each value
discrete analog The length of the bit sequence is a function of the number of
analog levels used in quantification. The sampling rate and the
number of bits per sample are two of the fundamental parameters to choose from
when you want to digitally process a certain audio signal.
Digital audio formats try to represent that set of samples
digital (or a modification) of them efficiently, so that it is optimized
depending on the application, either the volume of the data to be stored or the
processing capacity necessary to obtain the starting samples. In
in this sense there is a very extended audio format that is not considered audio
digital: the MIDI format. MIDI does not start with digital sound samples, but
stores the musical description of the sound, being a representation of the
score of them.
The digital audio system usually ends the reverse process to that described. From
the stored digital representation is obtained the set of samples that
represent. These samples go through a process of digital analog conversion
providing an analog signal that after processing (filtering,
amplification, equalization, etc.) affect the output transceiver (speaker)
which converts the electrical signal to a pressure wave that represents the sound.

Fundamental parameters of digital audio

The basic parameters to describe the sequence of samples it represents
The sound are:
ƒ The number of channels: 1 for mono, 2 for stereo, 4 for sound
quadraphonic, etc.
ƒ Sampling rate: The number of samples taken per second in each
channel.
ƒ Number of bits per sample: Usually 8 or 16 bits.
As a general rule, multichannel audio samples are usually organized in
frames A plot is a sequence of as many samples as channels,
each one corresponding to a channel. In this sense the number of samples per
second matches the number of frames per second. In stereo, the channel
Left is usually the first.