Human Hearing: An Approach to Compressing Audio Data


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Medical and physical examinations of human hearing and noise processing in the brain have shown that the hearing aid has its own perceptual characteristics. In certain circumstances, the brain does not register sounds or only partially registers them. Many signal components that are present in the acoustic signal are not even perceived by humans. The psychoacoustic call is concerned with investigating these facts. So far the following deficits in human ear perception have been discovered:

Curva auditiva del oído humano

Hearing perception range:

The waves can be emitted in a wide range of frequencies. However, the human ear can only really perceive a small section of this frequency range, the audio frequency range. In theory, humans can hear sounds with frequencies between 20 Hz and 20 kHz. In practice, however, it has been shown that ear sensitivity decreases considerably towards low and high frequencies. In the image above, amplitude, that is, sound pressure, is plotted against frequency.

Curva de audición específica de una pieza musical

Measurements have shown that all signals that are completely below the threshold of hearing at rest (red line) are inaudible. The amplitude of these tones (green peaks in the image) is too low, so their volume is too low to be perceived. It is interesting to see that the silent hearing threshold is not constant at a certain amplitude value, but changes with frequency. Very low tones (less than 50 Hz) are only perceptible from very high amplitudes, as are tones above 15 kHz. It should also be noted that not everyone has the same silent hearing threshold. Children can hear high frequencies much better than older people.

Masking:

Another deficit of the human hearing aid is the inability to distinguish between tones of very similar frequency and very different volume that occur simultaneously. This effect is also called auditory masking. Or German called simultaneous masking. A high-amplitude signal (dark blue in the image above), also known as a masker, hides quieter signals that have a similar frequency. In the image, these are all signals that are within the area highlighted in yellow. Some turquoise peaks are shown as an example. The yellow area is outlined by the orange individual masking threshold of the masker. The individual masking threshold and the silent hearing threshold can be combined to form the so-called global masking threshold. Thus, all signals below the global masking threshold are inaudible. In practice, auditory masking means nothing more than loud music signals cover the quiet parts and make them inaudible.
Another masking effect occurs when two tones follow each other in a very short time. Of these two tones, only the one with greater amplitude is perceived, that is, greater volume. Interestingly, even if the soft sound reaches the ear first, only the strong signal that arrives later is registered in the brain. This second important masking effect is also called temporary masking in technical jargon.

Low-frequency localization deficits:

Although the human ear is able to pinpoint the origin of high and mid-frequency tones in the room well, problems arise in the lower-frequency region. The brain calculates the location of the sound source from the difference in signal transit time between the left and right ears. If there is a sound source on the right, the waves emitted by this source are perceived earlier by the right ear than by the left. The origin of the tones is calculated from the time interval between the perception of the left and right ears. However, very low-frequency sound signals have very long wavelengths, making clear localization impossible. Therefore, there is practically no tonal difference between a mono sound source for low-frequency signals and a stereo sound source for very low-frequency sounds. Joint stereo effect. It is used, for example, in the construction of subwoofer satellite systems and is also a starting point for audio compression in the area of ​​low tones.
Therefore, the human ear can only improperly or not at all perceive a complete series of frequency ranges. In electrical engineering, the field of digital signal processing (DSP) deals, among other things, with mathematical processes that, in combination with the psychoacoustic model of the hearing aid, lead to data reduction.


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How much compresses an MP3

How much compresses an MP3

MP3 compression was an engineering response to the problem of digital storage and its large memory resource requirements. A conventional digital signal called PCM (Pulse Code Modulation) could easily require up to 10 Megabytes of memory per minute. This would represent about 30 Mb for a three minute song.
That requirement for storage memory could be handled by any computer if it were a few files, but when talking about three thousand songs the numbers become worrying. As if this were not enough, there is the problem of the Internet and its current transmission speeds. In the case of telephone lines, they have a limitation on their transmission bandwidth, so very large or heavy files represent a problem for conventional network traffic.

MPEG3 compression is considered the sound part of the original MPEG1 format that was intended for cinematography. Its abbreviations, Moving Picture Experts Group come from the committee that was created by the ISO Organization (international Standards Organization) and IEC ((International Electrotechnical Commission) to develop this format. Its principle is based on the Psychoacoustic model.

The human ear is known to discriminate sound according to its limitations. According to subject matter expert Paul Sellars, “If you hear solitary applause in a room, it will surely sound loud, but if it is preceded by the sound of a gunshot, it will sound fainter. The same thing happens in a room when you record a rock band, at a certain moment the strongest sound guitar in the mix, until the moment the drummer plays a certain cymbal, at which point the guitar will seem to attenuate “This phenomenon is used by the MP3 algorithm to perform its compression . I once explained it in the article that talked about ATRAC compression of the Minidisc.

The MP3 format divides the sound into 32 sub-bands, which allows it, according to the Psychoacoustic model on which it is based, to give priority to one element over another. At a certain moment in the material we can have a predominant low frequency sound of the kick drum, a high frequency of the cymbal and the vocalist at the same time. The algorithm is not that it eliminates two of them, but that it dedicates less storage space to them.

The mathematical part used with MP3 compression goes through the Shannon-Nyquist theorem, which states that for a wave to be properly reproduced in PCM digital format, its frequency of takes (Sampléo) must be twice the highest that is want to reproduce. In this case if we want to reproduce the frequency of 22.5KHz, (The auditory range oscillates between 20Hz-20KHz), our sampling frequency should be 44.1KHz.

The Fast Fourier Transform (FFT) is also used, which as we know can decompose a complex wave (PCM material) into a fundamental wave with its harmonics, all from its amplitude. The Discrete Cosine Transform is also used, which is based on the FFT but only using the real numbers

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These formats will continue to be perfected and emerge, but it should be understood that despite being disseminated there may be details that will not be perceived. In other words, for serious Audio work this format should not be used.

Some improvements can be made by looking for compressors that have a better ratio, such as 224, 256 and 320 Kbps. You can also consider using VBR (Variable Bit Rate) encoding where musical passages with greater dynamic complexity are treated with a higher rate. storage in contrast to the simplest. However, this will bring other complications because not all the reproducers can handle them.