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 … )

Continue reading A Basic Limitation in Stereo FM Reproduction

Threshold Elimination in Compressed Sound

I’ve written quite a few postings in this blog, about sound compression based on the Discrete Cosine Transform. And mixed in with my thoughts about that – where I was still, basically, trying to figure the subject out – were my statements to the effect that frequency-coefficients that are below a certain threshold of perceptibility could be set to zeroes, thus reducing the total number bits taken up, when Huffman-encoded.

My biggest problem in trying to analyze this is, the fact that I’m considering generalities, when in fact, specific compression methods based on the DCT, may or may not apply threshold-elimination at all. As an alternative, the compression technique could just rely on the quantization, to reduce how many bits per second it’s going to allocate to each sub-band of frequencies. ( :1 ) If the quantization step / scale-factor was high enough – suggesting the lowest quality-level – then many coefficients could still end up set to zeroes, just because they were below the quantization step used, as first computed from the DCT.

My impression is that the procedure which gets used to compute the quantization step remains straightforward:

  • Subdivide the frequencies into an arbitrary set of sub-bands – fewer than 32.
  • For each sub-band, first compute the DCTs to scale.
  • Take the (absolute of the) highest coefficient that results.
  • Divide that by the quality-level ( + 0.5 ) , to arrive at the quantization step to be used for that sub-band.
  • Divide all the actual DCT-coefficients by that quantization step, so that the maximum, (signed) integer value that results, will be equal to the quality-level.
  • How many coefficients end up being encoded to having such a high integer value, remains beyond our control.
  • Encode the quantization step / scale-factor with the sub-band, as part of the header information for each granule of sound.

The sub-band which I speak of has nothing to do with the fact that additionally, in MP3-compression, the signal is first passed through a quadrature filter-bank, resulting in 32 sub-bands that are evenly-spaced in frequencies by nature, and that the DCT is computed of each sub-band. This latter feature is a refinement, which as best I recall, was not present in the earliest forms of MP3-compression, and which does not affect how an MP3-file needs to be decoded.

(Updated 03/10/2018 : )

Continue reading Threshold Elimination in Compressed Sound

Emphasizing a Presumed Difference between OGG and MP3 Sound Compression

In this posting from some time ago, I wrote down certain details I had learned about MP3 sound compression. I suppose that while I did write, that the Discreet Cosine Transform coefficients get scaled, I may have missed to mention in that same posting, that they also get quantized. But I did imply it, and I also made up for the omission in this posting.

But one subject which I did mention over several postings, was my own disagreement with the practice, of culling frequency-coefficients which are deemed inaudible, thus setting those to zero, just to reduce the bit-rate in one step, hoping to get better results, ‘because a lower initial bit-rate also means that the user can select a higher final bit-rate…’

In fact, I think that some technical observers have confused two separate processes that take place in MP3:

  1. An audibility threshold is determined, so that coefficients which are lower than that are set to zero.
  2. The non-zero coefficients are quantized, in such a way that the highest of them fits inside a fixed maximum, quantized value. Since a scale-factor is computed for one frequency sub-band, this also implies that close to strong frequency coefficients, weaker ones are just quantized more.

In principle, concept (1) above disagrees with me, while concept (2) seems perfectly fine.

And so based on that I also need to emphasize, that with MP3, first a Fast-Fourier Transform is computed, the exact implementation of which is not critical for the correct playback of the stream, but the only purpose of which is to determine audibility thresholds for the DCT transform coefficients, the frequency-sub-bands of which must fit the standard exactly, since the DCT is actually used to compress the sound, and then to play it back.

This FFT can serve a second purpose in Stereo. Since this transform is assumed to produce complex numbers – unlike the DCT – it is possible to determine whether the Left-Minus-Right channel correlates positively or negatively with the Left-Plus-Right channel, regarding their phase. The way to do this effectively, is to compute the dot-product between two complex numbers, and to see whether this dot-product is positive or negative. The imaginary component of one of the sources needs to be inverted for that to work.

But then negative or positive correlation can be recorded once for each sub-band of the DCT as one bit. This will tell, whether a positive difference-signal, is positive when the left channel is more so, or positive if the right channel is more so.

You see, in addition to the need to store this information, potentially with each coefficient, there is the need to measure this information somehow first.

But an alternative approach is possible, in which no initial FFT is computed, but in which only the DCT is computed, once for each Stereo channel. This might even have been done, to reduce the required coding effort. And in that case, the DCT would need to be computed for each channel separately, before a later encoding stage decides to store the sum and the difference for each coefficient. In that case, it is not possible first to determine, whether the time-domain streams correlate positively or negatively.

This would also imply, that close to strong frequency-components, the weaker ones are only quantized more, not culled.

So, partially because of what I read, and partially because of my own idea of how I might do things, I am hoping that OGG sound compression takes this latter approach.

Dirk

Continue reading Emphasizing a Presumed Difference between OGG and MP3 Sound Compression