An observation about some purchased FLAC Files.

One of the ideas which I’ve blogged about often – a pet peeve of mine – is how lossy compression is not inaudible, although some people have claimed it is, and how its use degrades the final quality of modern, streamed or downloaded music.

And so if this is taken to be real for the moment, a question can rise as to what the modern methods are, to purchase High-Fidelity, Classical Music after all. One method could be, only to purchase Audio CDs that were mastered in the 1990s. But then, the eventual problem becomes, that even the best producers may not be mastering new recordings in that format anymore, in the year 2019. We may be able to purchase famous recordings made in the 1990s, but none from later, depending on what, exactly, our needs are. But, an alternative method exists to acquire such music today, especially to acquire the highest quality of Classical music recorded recently.

What people can do is to purchase and download the music in 16-bit, FLAC-compressed format. Ideally, this form of compression should not insert any flaws into the sound on its own. The sound could still be lacking in certain ways, but if it is, then this will be because the raw audio was flawed, before it was even compressed. By definition, lossless compression decompresses exactly to what was present, before the sound was compressed.

I have just taken part in such a transaction, and downloaded Gershwin’s Rhapsody In Blue, in 16-bit FLAC Format. But I made an interesting observation. The raw 16-bit audio at a sample-rate of 44.1kHz, would take up just over 1.4mbps. When I’ve undertaken to Flac-compress such recordings myself, I’ve never been able to achieve a ratio much better than 2:1. Hence, I should not be able to achieve bit-rates much lower than 700kbps. But the recording of Gershwin which I just downloaded, achieves 561kbps. This is a piece in which a piano and a clarinet feature most prominently, and, in this version, also some muted horns. And yet, the overall sound quality of the recording seems good. So what magic might be employed by the producers, to result in smaller FLAC Files?

(Updated 8/19/2019, 17h00 … )

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A Basic Limitation in Stereo FM Reproduction

One of the concepts which exist in modern, high-definition sound, is that Human Sound perception can take place between 20 Hz and 20kHz, even though those endpoints are somewhat arbitrary. Some people cannot hear frequencies as high as 20kHz, especially older people, or anybody who just does not have good hearing. Healthy, young children and teenagers can typically hear that entire frequency range.

But, way back when FM radio was invented, sound engineers had flawed data about what frequencies Humans can hear. It was given to them as data to work with that Humans can only hear frequencies from 30Hz to 15kHz. And so, even though Their communications authorities had the ability to assign frequencies somewhat arbitrarily, they did so in a way that was based on such data. (:1)

For that reason, the playback of FM Stereo today, using household receivers, is still limited to an audio frequency range from 30Hz to 15kHz. Even very expensive receivers will not be able to reproduce sound, that was once part of the modulated input, outside this frequency range, although other reference points can be applied, to try to gauge how good the sound quality is.

There is one artifact of this initial standard which was sometimes apparent in early receivers. Stereo FM has a pilot frequency at 19kHz, which a receiver needs to lock an internal oscillator to, but in such a way that the internal oscillator runs at 38kHz, but such that this internal oscillator can be used to demodulate the stereo part of the sound. Because the pilot signal which is actually part of the broadcast signal is ‘only’ at 19kHz, this gives an additional reason to cut off the audible signal at ‘only’ 15Khz; the pilot is not meant to be heard. But, way back in the 1970s and earlier, Electrical Engineers did not have the type of low-pass filters available to them which they do now, that are also known as ‘brick-wall filters’, or filters that attenuate frequencies above the cutoff frequency very suddenly. Instead, equipment designed to be manufactured in the 1970s and earlier, would only use low-pass filters with gradual ‘roll-off’ curves, to attenuate the higher frequencies progressively more, above the cutoff frequency by an increasing distance, but in a way that was gentle. And in fact, even today the result seems to be, that gentler roll-off of the higher frequencies, results in better sound, when the quality is measured in other ways than just the frequency range, such as, when sound quality is measured for how good the temporal resolution, of very short pulses, of high-frequency sound is.

Generally, very sharp spectral resolution results in worse temporal resolution, and this is a negative side effect of some examples of modern sound technology.

But then sometimes, when listeners with high-end receivers in the 1970s and before, who had very good hearing, were tuned in to an FM Stereo Signal, they could actually hear some residual amount of the 19kHz pilot signal, which was never a part of the original broadcast audio. That was sometimes still audible, just because the low-pass filter that defined 15kHz as the upper cut-off frequency, was admitting the 19kHz component to a partial degree.

One technical accomplishment that has been possible since the 1970s however, in consumer electronics, was an analog ‘notch filter’, which seemed to suppress one exact frequency – or almost so – and such a notch filter could be calibrated to suppress 19kHz specifically.

Modern electronics makes possible such things as analog low-pass filters with a more-sudden frequency-cut-off, digital filters, etc. So it’s improbable today, that even listeners whose hearing would be good enough, would still be receiving this 19kHz sound-component to their headphones. In fact, the sound today is likely to seem ‘washed out’, simply because of too many transistors being fit on one chip. And when I just bought an AM/FM Radio in recent days, I did not even try the included ear-buds at first, because I have better headphones. When I did try the included ear-buds, their sound-quality was worse than that, when using my own, valued headphones. I’d say the included ear-buds did not seem to reproduce frequencies above 10kHz at all. My noise-cancelling headphones clearly continue to do so.

One claim which should be approached with extreme skepticism would be, that the sound which a listener seemed to be getting from an FM Tuner, was as good as sound that he was also obtaining from his Vinyl Turntable. AFAIK, the only way in which this would be possible would be, if he was using an extremely poor turntable to begin with.

What has happened however, is that audibility curves have been accepted – since the 1980s – that state the upper limit of Human hearing as 20kHz, and that all manner of audio equipment designed since then takes this into consideration. This would include Audio CD Players, some forms of compressed sound, etc. What some people will claim in a way that strikes me as credible however, is that the frequency-response of the HQ turntables was as good, as that of Audio CDs was. And the main reason I’ll believe that is the fact that Quadraphonic LPs were sold at some point, which had a sub-carrier for each stereo channel, that differentiated that stereo channel front-to-back. This sub-carrier was actually phase-modulated. But in order for Quadraphonic LPs to have worked at all, their actual frequency response need to go as high asĀ  40kHz. And phase-modulation was chosen because this form of modulation is particularly immune to the various types of distortion which an LP would insert, when playing back frequencies as high as 40kHz.

About Digital FM:

(Updated 7/3/2019, 22h15 … )

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Butterworth Filters

There exists a basic type of low-pass filter, called a Butterworth Filter, which is a 2nd-order filter, which therefore has a falloff-rate of -12db /Octave, far above the corner frequency, and this is its general diagram:

basic_1

Even though it is clear from this diagram that the two capacitors, or the two resistors, are allowed to have different values, the way the design of this filter is mainly taught today, both resistors are made equal, as are both capacitors, thus simplifying the computation of each, once the other has been determined according to what seems practical, applying the same principle as what would be applied for a 1st-order filter.

One basic weakness of this filter, especially in modern applications, is the fact that it will attenuate frequency-components considerably, which are below its corner-frequency. There have historically been two approaches taken to reduce this effect, if any attempt has been made to do so at all:

  1. C1 can be given twice the value of C2, but R1 and R2 kept equal. This poses the question of whether the corner-frequency will still be correct. And my estimation is that because of the way Electrical Engineers have defined the corner-frequency, the specific frequency-response at that frequency should remain the square root of 1/2 (or, -3db). But, if C1 is larger than C2, then the frequency-response will not be the same at any other point in the curve. I.e., the curve could be flatter, with response-values closer to unity, at frequencies considerably lower than the corner-frequency.
  2. The operational amplifier stage, which in the basic design is just a voltage-follower, can be transformed into a gain-stage, with a gain slightly higher than one. This is done by placing a voltage-divider from the output of an operational amplifier, to yield the feedback voltage, fed to its inverting input. What needs to be stressed here, is that significantly high gain leads to an unstable circuit.

While either approach can be taken, it is important not to apply both at the same time, as the amount of feedback given by C1 would be exaggerated, and would lead to a hot-spot somewhere in the pass-band of this filter. In general, the trend today would be to use approach (2).

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About The Applicability of Over-Sampling Theory

One fact which I have described in my blog, is that when Audio Engineers set the sampling rate at 44.1kHz, they were taking into account a maximum perceptible frequency of 20kHz, but that if the signal was converted from analog to digital format, or the other way around, directly at that sampling rate, they would obtain strong aliasing as their main feature. And so a concept which once existed was called ‘over-sampling’, in which then, the sample-rate was quadrupled, and by now, could simply be doubled, so that all the analog filters still have to be able to do, is suppress a frequency which is twice as high, as the frequencies which they need to pass.

The interpolation of the added samples, exists digitally as a low-pass filter, the highest-quality variety of which would be a sinc-filter.

All of this fun and wonderful technology has a main weakness. It actually needs to be incorporated into the devices, in order to have any bearing on them. That MP3-player, which you just bought at the dollar-store? It has no sinc-filter. And therefore, whatever a sinc-filter would have done, gets lost on the consumer.

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