|
Arabic
Bengali
Chinese
English
French
German
Hebrew
Hindi
Italian
Japanese
Korean
Malay
Polish
Portuguese
Spanish
Turkish
Ukrainian
Vietnamese|
BOOKS AND ARTICLES
The development of surround sound systems - from mono to 3D
At present, two-channel stereophony has already become a classic way of transmitting and reproducing sound. The purpose of stereophonic sound reproduction is to reproduce the sound image as accurately as possible. Sound localization is only a means to get a richer and more natural sound. However, the transmission of spatial information by the most common "classic" two-channel systems has a number of disadvantages, which encourages designers to create various surround sound systems. The listener in the concert hall hears not only the direct sound coming from individual instruments of the orchestra, but also coming from different directions (including from behind), diffused (diffuse) sound reflected from the walls and ceiling of the room, which creates the effect of space and completes the overall impression. The delay with which diffuse sound reaches the listener's ears and its spectral composition depend on the size and acoustic properties of the room. In a two-channel transmission, the information created by the diffuse sound is largely lost, and in the case of a studio recording, it may not be present initially. The human ear best localizes sound sources in the horizontal plane. At the same time, sounds coming from behind, in the absence of additional information, are localized worse. Vision, including peripheral vision, is the main sense of determining the location of objects, therefore, without visual information, the ability to assess the position of sound in the vertical plane and its distance from us is weak and rather individual. In part, this can be explained by the individual anatomical features of the auricles. When playing records, visual information is absent, so any sound technology for the mass market that claims to be "surround sound" is forced to create something average and deliberately compromise. Many methods can be used to reproduce or synthesize the "hall effect". Back in the mid-50s, Philips, Grundig, Telefunken tested 3D and Raumton three-dimensional playback systems. The sound transmission was monophonic, but additional loudspeakers (usually built-in, less often remote), radiating sound sideways or upwards, created the impression of a large space due to the sound reflected from the walls and ceiling. Since the echo signal delay in domestic premises is rather small, spring reverberators were later used in the channel for amplifying additional signals to increase it. These systems, due to the technical complexity that was significant for that time, did not last long on the market and quickly left the stage. Subsequently, ambiophonic systems were developed to transmit diffuse sound, which were used mainly in cinema. The additional channel (or channels) for transmitting diffuse sound in such systems have less power than the main ones, and their frequency range corresponds to the diffuse signal frequency band (approximately 300...5000 Hz). Radiation of additional speakers should be diffused, for which they are directed to the walls or ceiling of the listening room.
The complexity of standardization and technical problems with recording and transmitting signals of three, four or more channels led to the fact that two-channel stereophony became the main system for recording and transmitting sound for many years. But attempts to create surround sound systems did not stop. The development of ambiophony was quadraphony (four-channel sound reproduction), which peaked in popularity in the first half of the 70s. In contrast to the ambiophonic system, here all sound reproduction channels are equipped equally. Discrete (full) quadraphony, which provides the maximum effect of presence, requires four channels of sound transmission and, therefore, turned out to be incompatible with the technical means of sound recording and broadcasting that existed at that time.
To overcome this obstacle, several systems of matrix quadraphony (in the terminology of that time - quasi-quadraphony) were created, in which the original signals of four channels were matrixed for transmission over two channels, and during playback, the original signals were restored by sum-difference transformations, and without a decoder it was possible to reproduce normal stereo signal. Since none of these systems were either fully quad or fully compatible with two-channel stereophony due to the high signal penetration from channel to channel, their practical application was limited and interest in them quickly faded.
There were no winners in the "war of standards" of quad systems, the idea died safely, the principles were forgotten, but the term remained. Therefore, now few people are embarrassed by the fact that "something" that has four amplification channels and four speakers is proudly called a "quad system". However, this is fundamentally wrong, since the signal source remains two-channel, and the signals of the front and rear channels with this system construction differ from each other only in level, that is, the panning principle is used. Panning has been widely used in stereo production since the mid-50s to position monophonic audio signals "left/right/middle" of the sound field. When panning, there is no effect on the frequency and phase of the signal, only the level of the mono signal supplied to each of the stereo channels changes. Panning to multiple channels (in the case of multi-channel recordings) is done in a similar way. However, when determining the direction to the sound source, our hearing aid uses not only the difference in the intensity of sound signals, but also the phase shift between them, and the effect of the phase shift on the accuracy of sound source localization is most pronounced in the frequency range from approximately 500 to 3000 Hz. (Again, the frequency range of diffuse sound!). Therefore, simple panning does not provide the desired sound fidelity. Stereo effects ("running sound", "left-right" sound binding, etc.) of the first stereo recordings quickly became boring. Therefore, the best recordings of electronic instruments in the studio in the 60s were made using microphone technology, which explains the "live" nature of the sound: The introduction of multi-channel fully electronic (without the use of microphones) recording of instruments with subsequent mixing, facilitating the work of the sound engineer, at the same time destroyed the atmosphere of the hall. Subsequently, this fact began to be taken into account when conducting studio recordings, although there was no complete return to microphone technology. When using a two-channel playback scheme, the main zone of effective location of apparent sound sources (PSZ) is located in front of the listener and covers a space of about 180 degrees in the horizontal plane. The two front channels are not able to adequately reproduce sounds whose sources are in reality located behind and in a vertical plane, if there is no support in the form of additional signals. The use of rear speakers in combination with sound panning works well with the sound sources in front and behind the listener and weaker with the side position. However, sound panning alone will never provide an acceptable positioning of sound sources in the vertical plane. During the development of matrix systems, it turned out that a significant part of the spatial information is contained in the difference signal (stereo information signal), which can be fed to the rear channel loudspeakers either in pure form or mixed with a certain proportion of front signals. In the simplest case, this does not even require additional amplification channels, and signals can be matrixed at the amplifier output:
This is how several pseudo-quadraphonic systems were born, which completely ousted the "true Aryans" from the market in the mid-70s. They differed from each other only in the ways of obtaining a difference signal. However, their triumph was also short-lived, which was explained by the shortcomings of the signal carrier - the vinyl disc and magnetic tape. The uncorrelated noise of the left and right channels was not subtracted, which, in combination with the relatively low level of the difference signal, greatly worsened the signal-to-noise ratio in the rear channels. Another, no less significant drawback of such systems is the lack of dependence of the rear signal level on the nature of the phonogram. With a low level of the rear signal, the spatial effect is hardly noticeable, with an increase in the level, a break in the sound stage appears and its fragments move backwards (the effect of "surrounding by an orchestra", which does not correspond to reality). When playing back "live" recordings (having a natural distribution of sum, difference and phase components), this disadvantage was insignificant, but on most studio phonograms, the rear channels introduced significant errors in the position of the CIP. To overcome this shortcoming, early surround sound systems tried to use automatic panning. Control signals were obtained from the level of spatial information - an increase in the level of difference signals led to an increase in gain in the rear channels. However, the adopted panning model was very crude, as a result of which the expander control errors led to a chaotic change in the level of the rear signals (the "heavy breathing" effect). Interest in surround sound systems has re-emerged with the advent of digital media, the level of their own noise is negligible, and even analog signal processing will hardly degrade the dynamic range of the system. The development of digital signal processing methods led to the creation of digital sound processors (Digital Sound Processor - DSP). Originally developed for home theater systems, surround sound processors have recently begun to be actively used in car audio systems. Their use can significantly improve the sound in the car, so they are produced not only in the form of separate DSP-devices, but are also part of relatively inexpensive radio tape recorders. Processor settings allow you to select the most optimal parameters for the selected listening position. There are a number of methods that allow equipment to reproduce sound localized in space with a limited number of speakers. Different implementation methods have strengths and weaknesses, so it is important to understand the fundamental differences between the main signal processing methods. At the heart of modern surround sound systems (Dolby Surround, Dolby Pro-Logic, Q-Sound, Curcle Surround and others) is the same idea of sum-difference conversion, supplemented by "proprietary" signal processing methods (both analog and digital). Often they are united by the common name "3D-systems" ("rebirth" of the term forty years ago!). Before looking at the principles involved in processing audio signals in surround sound systems, let's recap the typical recording process. First, a recording is made that has many individual channels—instruments, voices, sound effects, and so on. During mixing, for each audio track, the volume level and the location of the audio source are controlled to achieve the desired result. In the case of stereo recording, the result of mixing is two channels, for surround systems the number of channels is higher (for example, 6 channels for the Dolby Digital/AC-5.1 "3" format). In any case, each channel consists of signals that are intended to be routed to individual speakers when the user listens. Each of these signals is the result of complex mixing of the original source signals. Next, the process of encoding the channels received after mixing takes place, and as a result one digital stream (bitstream) is obtained. During playback, the decoder processes the digital stream, dividing it into individual channels and transmitting them for playback to speaker systems. For multi-channel (discrete) surround sound systems, it is possible to simulate real-life acoustic systems (Phantom mode). If you only have two speakers, then the subwoofer (woofer) and center (dialog) channels are simply added simultaneously to both output channels. The rear left channel is added to the left output channel, the rear right channel to the right output channel. Recall that panning only affects the amplitude of the audio signal. Audio conversion in modern 3D systems includes additional information about the amplitude and phase difference/delay between the output channels in the audio stream. Typically, the amount of processing depends on the frequency of the signal, although some effects are created using simple time delays. What methods are used to process the audio signal? First of all, this is the expansion of the stereo base (Stereo Expansion), which is produced by influencing the differential stereo signal of the front channels. This method can be considered classical and it applies primarily to conventional stereo recordings. Signal processing can be either analog or digital. Secondly, Positional 3D Audio (localized 3D sound). This method operates on many individual audio channels and attempts to individually locate each signal in space. Thirdly, Virtual Surround (virtual surround sound) is a method of playing multi-channel recording using a limited number of sound sources, for example, playing five-channel sound on two speakers. It is obvious that the last two methods are applicable only to multichannel audio media (recordings in DVD, AC-3 format), which is not very important for automotive systems so far. Rounding out the list are various methods of artificial reverb. As sound propagates through space, it can be reflected or absorbed by various objects. Reflected sounds in a large space can actually create a clearly distinguishable echo, but in a limited space there is a combination of many reflected sounds so that we hear them as a single sequence that follows the original sound and fades, and the degree of attenuation is different for different frequencies and directly depends from the properties of the environment. Digital sound processors use a generalized reverberation model, which reduces the control of the reverberation process to setting key parameters (delay time, number of reflections, decay rate, change in the spectral composition of reflected signals). Thus, the hall, live, stadium, etc. modes are implemented. The simulation is quite realistic. Analog processors use signal delay lines for this purpose. The control of the reverb parameters in this case is much more complicated, so there is usually only one fixed mode of operation. Of course, it is difficult to describe the structural features of all existing surround sound systems, but their work is based on the principles considered - the difference is only in the details of the algorithms and the set of modes (presets). Therefore, the best adviser when choosing a sound processor is your own hearing. Publication: www.bluesmobil.com/shikhman
▪ How to make a small box big or something about padding ▪ Combined frequency response control unit
Artificial leather for touch emulation
15.04.2024 Petgugu Global cat litter
15.04.2024 The attractiveness of caring men
14.04.2024
▪ A new way to obtain alternative fuel ▪ Silver nanothreads keep you warm ▪ Christmas decorations make us happier
▪ section of the site Wonders of nature. Article selection ▪ article There is something to despair from. Popular expression ▪ article Grapes virginian. Legends, cultivation, methods of application ▪ article Fixing AVR fuzzes. Encyclopedia of radio electronics and electrical engineering ▪ article Missing card. Focus Secret
Home page | Library | Articles | Website map | Site Reviews
www.diagram.com.ua |
We recommend interesting articles Section 
See other articles Section 
Latest news of science and technology, new electronics:
All languages of this page