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Why does even digital audio deteriorate? Part 2

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Why does even digital audio deteriorate? Part 2

Digital Audio

I am not an audiophile, and I am not the type that is very demanding to listen, which is why I am not aware of so-called pure audio.
So I didn’t know Mr. Kanai at all, but he seems to be famous for that source.

Digital Audio

The reason I met Mr. Kanai was because I saw the serialized article “What is the definitive SACD born in the” Kaimaru Room “” from the “Ken Fujimoto Weekly Digital Audio Lab” which I have long subscribed to ? , this article was really interesting.

This is an interview article about the production process of Emi Fujita’s (Le Couple) work “Manzanilla Best Audio”, but it is very easy to understand the difference in mindset between the production side and the actual listener. I think .

Anyway, the content on Mr. Kanai’s HP was scaled content for me.
It’s a good opportunity, so I’d like to change my mind a bit.

Especially around surround sound, you need to study.
I cannot understand it at all because I have not tried surround sound as a real experience.

I am also very interested in SACD, but I am very concerned about buying a PS3 because I do not have a playback environment.

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Why does even digital audio deteriorate?

Digital Audio

It is not limited to DTM and DAW, I think if you are a musician you may have noticed the deterioration in sound quality.

Digital Audio

For example, change the shield to a higher one or allow it to be bypassed entirely when the effector is not in use.
When it comes to old stories, record without ping-pong as much as possible.

I don’t think the deterioration in sound quality bothers me, but I’m obviously not afraid of losing sound, so I’m careful.
But, it is simply an analog of the story in, don’t use your mind as I don’t say anything about digital audio.

Why?
That’s because I couldn’t fully understand the concept of “digital data degradation”.

When it comes to guitars, it’s easy to see that upgrading the various effectors and protectors between the guitar and the amp, and the protector that goes to the amp’s audio I / O “improves the sound.”
It is an analog signal.
But I couldn’t quite understand the history of changing the Firewire cable connecting the audio I / O to the PC to improve the sound quality.

It does not matter if it is via the Internet or copying from a medium, but when you think about it normally and transfer data digitally, there is no deterioration.
To be precise, transmission loss always occurs, so the signal itself deteriorates, but when the data of the transfer result is considered as the center, the picture is that the transfer retries increase rather than deteriorate, and on the user side. From the point of view, I don’t think it can be said that the transfer time has increased and the data has deteriorated.

If the transmission loss is very large, the file itself may be corrupted, and in the case of data to be processed in real time, the transfer may not be on time and the processing may result in an error, but it is transfers normally. In that case, I thought it was digital data that the data should be the same before and after the transfer, no matter how much transmission loss occurred or how long it would take …
(This is just my own expectation. I don’t know if it fits).

Also, in terms of sound, there are two patterns: deterioration of the analog sound quality, which is literally “deterioration” that produces sloppy sound like “thinning sound”, and noise mixed in the transmission path. I think that in the case of a digital data error, it is not a level that says “the sound is bad”, but it becomes a choppy sound or a loud sound that can only be called noise.

Even in digital, analog affects sound quality

analog digital

Audio network audio for PC

analog digital audio

Whether you listen to music or watch videos on television, it is becoming more and more common to use digital data as a sound source.

With the improvement of the quality of communications, such as optical lines on the Internet, the amount of information is increasing and the enjoyment and choices for users are increasing.

However, whether you listen to music on a smartphone or PC audio, the sound quality of subscriptions differs by high resolution, but analog is really important to fully bring out the high quality of the source of sound.

Analog opinions that are not anti-digital

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Analog opinions that are not anti-digital
At first glance, the difference in the amount of digital information appears to be the deciding factor.
CD player whose analog circuit influences the sound.

At first glance, the difference in the amount of digital information appears to be the deciding factor.

Digital sound sources (software) that started with CDs are now changing for downloads and stories.

Music data has an Internet environment and digital devices, such as PCs, network players, and transmitters, receive digital signals and a DA converter converts them to analog.

The analog signal converted from digital is amplified by the amplifier and sound is output from the speaker.

Recently, it seems that the digital sound source in the smartphone is popular for products that play music directly from the speaker via Wi-Fi or Bluetooth, but in fact, the Bluetooth speaker has a DA converter that converts digital to analog. ..

How is the digital representation of signals different from analog?

How is the digital representation of signals different from analog?

analog and digital

The traditional analog representation of signals is based on the similarity (similarity) of electrical signals (changes in current and voltage) with the original signals represented by them (sound pressure, temperature, speed, etc.).

Analog vs Digital

As well as in the similarity of the forms of the electrical signals in various points of the amplification or transmission path. The shape of the electrical curve that describes (also called transfer) the original signal is as close as possible to the shape of the curve of this signal.

Such a representation is the most accurate, however, the slightest distortion of the shape of the electrical carrier signal will inevitably involve the same distortion of the shape and signal of the carrier. In terms of information theory, the amount of information in the carrier signal is exactly equal to the amount of information in the original signal, and the electrical representation does not contain redundancy that could protect the carried signal from distortion during storage, transmission. and amplification.

The digital representation of electrical signals is designed to add redundancy to avoid unwanted interference. For this, serious restrictions are imposed on the carrier electrical signal: its amplitude can take only two limit values: 0 and 1. In this case, the entire zone of possible amplitudes is divided into three zones: the lower one represents zero values, the upper , unique, and the intermediate is prohibited, inward. only interference can get in. Therefore, any interference whose amplitude is less than half the amplitude of the carrier signal does not affect the correct transmission of values 0 and 1. Interference with a higher amplitude also does not affect whether the duration of the interference pulse is significantly less than the duration of the information pulse.

The digital signal formed in this way can carry any useful information that is encoded in the form of a sequence of bits: zeros and ones; electrical and sound signals are a particular case of such information. Here, the amount of information in the digital carrier signal is much higher than in the original encoded signal, so the carrier signal has a certain redundancy with respect to the original, and any distortion of the waveform of the carrier signal, which still retains the receiver’s ability to correctly distinguish between zeros and ones, does not affect the reliability of the transmitted signal. by this information signal. However, in the case of exposure to significant interference, the shape of the signal can be distorted to such an extent that the precise transmission of the information being transferred becomes impossible: errors appear in it, which, with a simple coding method , the receiver can not only correct, but also detect. To further increase the resistance of a digital signal to interference and distortion, two types of digital redundancy coding are used: verification codes (EDC – Error Detection Code) and correction (ECC – Error Correction Code). ). Digital encoding is simply adding extra bits to the original information and / or converting the original bit string into a longer string and other structure. EDC allows you to simply detect the fact of an error: a distortion or loss of a useful one or the appearance of a false digit, but the information that is transferred in this case is also distorted; ECC allows you to immediately correct detected errors, keeping the information that is transferred unchanged.

Each type of EDC / ECC has its own capacity limit to detect and correct errors, after which undetected errors and distortions of the transferred information begin again. An increase in the volume of EDC / ECC relative to the volume of the original information generally increases the detection and correction capabilities of these codes.

Like EDC, the popular cyclic redundancy code CRC (Cyclic Redundancy Check), whose essence is the complex mixing of the initial information in the block and the formation of short binary words, whose bits have a strong cross-dependence on each bit of the block. Changing even one bit in a block causes a significant change in the CRC calculated from it, and the probability of such a bit distortion where the CRC does not change is extremely small even with short CRC words (a small percentage of the length of the block). The ECC uses the Hamming and Reed-Solomon codes, which also include EDC functions.

The information redundancy of the digital carrier signal leads to a significant expansion (by an order of magnitude or more) of the frequency band required for its successful transmission.

Analog-Digital Processing

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.