Myths of Digital Music Part 5
Myth 1
The wider the spectrum, the better the recording (over spectrograms, auCDtect and frequency range)
Today in forums, unfortunately, it is very common to measure the quality of a track with a “ruler in the spectrogram”. Obviously, for the simplicity of this method. But, as practice shows, in reality everything is much more complicated.
And the point is this. The spectrogram visually demonstrates the distribution of signal power at frequencies, but cannot give a complete picture of the sound of the recording, the presence of distortions and compression artifacts in it. That is, in fact, all that can be determined from the spectrogram is the frequency range (and partially, the density of the spectrum in the HF region). That is, in the best case, by analyzing the spectrogram, you can identify the upconversion. The comparison of the spectrograms of the tracks obtained by encoding several encoders with the original is absolutely absurd. Yes, you can identify differences in the spectrum, but determining whether (and to what extent) they will be perceived by the human ear is almost impossible. We must not forget that the task of lossy encoding is to provide a result that the human ear cannot distinguish from the original (not with the naked eye).
The same applies to evaluating the encoding quality by analyzing the output tracks with the auCDtect program (Audiochecker, auCDtect Task Manager, Tau Analyzer, fooCDtect are just shells for the one-of-a-kind auCDtect console program). The auCDtect algorithm also analyzes the frequency range and only allows you to determine (with a certain degree of probability) whether MPEG compression was applied in any of the encoding stages. The algorithm is designed for MP3, so it is easy to “cheat” with the Vorbis, AAC and Musepack codecs, so even if the program writes “100% CDDA”, it does not mean that the encoded audio is 100% identical to the original. .
And going straight back to the specters. Also popular is the desire of some “enthusiasts” to turn off the low pass filter on the LAME encoder at all costs. There is a lack of understanding of coding and psychoacoustic principles. First, the encoder cuts the high frequencies for one purpose: to save data and use it to encode the most audible frequency range. The extended frequency range can be fatal to overall sound quality and cause audible coding artifacts. Also, turning off the cutoff at 20 kHz is generally not justified, as a person simply does not hear the higher frequencies.
There is a kind of “magic” EQ preset that can significantly enhance the sound.
This is not entirely true, in the first place, because each configuration taken separately (headphones, acoustics, sound card) has its own parameters (in particular, its amplitude-frequency characteristic). And therefore each configuration must have its own unique approach. Simply put, such an EQ preset exists, but it is different for different settings. Its essence lies in adjusting the frequency response of the path, that is, in “leveling out” unwanted voltage dips and surges.
Also, among people who are far from direct work with sound, it is very popular to set the graphic equalizer “with a tick”, which actually represents an increase in the level of the low and high frequency components, but at the same Time leads to muffled vocals and instruments, whose sound spectrum is in the mid-range region.
Before converting music to another format, you must “unzip” it to WAV
I would like to point out right away that WAV stands for PCM (pulse code modulation) data in a WAVE container (file with extension * .wav). This data is nothing more than a sequence of bits (zeros and ones) in groups of 16, 24 or 32 (depending on the bit depth), each of which is a binary code of the corresponding sample width (for For example, for 16 bits in decimal notation (these are values from -32768 to +32768).
So the fact is that any sound processor, be it a filter or an encoder, generally works only with these values, that is, only with uncompressed data. This means that to convert audio from, say, FLAC to APE, you just need to decode FLAC to PCM first and then encode PCM to APE. It’s like repackaging files from ZIP to RAR, you need to unzip the ZIP first.
However, if you’re using a converter or just an advanced console encoder, intermediate to PCM conversion happens on the fly, sometimes even without writing to a temporary WAV file.
