General sound developments in radio, and other areas, question.

I'm a classically trained musician (and teacher) and enjoy listening to music for fun when I don't have to analyze/study it.

Since I'm new to older electronics, I am starting to notice some interesting tonal anomalies between lows and highs, also that tone control, even unrestored with my RCA Livingston with original caps (shame, I know) sounds great. I've actually used it as an example in my music production courses just to offer a comparison of different technologies.

I have learned that amplification and balancing highs and lows was something multiple companies were working towards, and the hi-fi developments removed the 'dead' spots and increased the range of frequency response in the output area. Isn't that what golden throat and other notable names for that idea attempted? I think I can hear the clarity that the golden throat (whatever it actually means) does to human speaking voices, but what should it do for music?

*Does having to work with monophonic output create a lack of depth during the production phase, and is it noticeable? I feel like I can hear it but I could also be plugging in missing detail due to my ear training.

*What I mean to say is that I perceive depth, and can actually hear what I would consider a face-first soundscape, since it is monophonic; I'm going to enjoy restoring the Livingston to compare and contrast tone and depth.

Additionally - that 1966 new vista console with am/fm phono has one heck of an output at even volume level 1.5 out of 9, it's too big for my apartment in the sonic respect.

[Titan1a]

GOLDEN THROAT is just advertising. I've seen that golden sticker on radios that were no different than any other units. There were manufactures of very high fidelity radios from the '30's. They weren't cheap either! The one that stands out for rarity is the Zenith Stratosphere. Crosby made a rare model of only three radios that included every known technology and it was huge and heavy! These are only a sample.

[old_tv_nut]

This is an interesting topic to me. I think professional musicians can be poor judges of sound reproduction because they can fill in missing stuff in their heads - but that's just a wild surmise, because my musical experience consists of 4 years of playing second clarinet in the high school concert band.

As far as tonal balance of bass and treble in systems with limited bandwidth (like AM radio), there was a rough rule of thumb that the low frequency and high frequency cutoffs should be balanced, that is, a system without extended highs should also not have extended lows. I think there was even a rough calculation that the product of the low frequency cutoff and high frequency cutoff should be roughly a certain value, but I don't recall the details. I don't know how this affects old guys like me that are losing their high frequency sensitivity, but not their bass sensitivity. We are probably doing a mental (psychophysical) compensation and just hearing less of the difference between AM radio and HiFi.

I also want to mention that some distortion of mid frequencies can generate synthetic highs that make a speaker system sound like it has better high frequency response than it really does (if it's not too extreme). This was common in the small speakers in TV sets. A side by side comparison with a system with real high frequency response will reveal the difference, however.

Thanks for the information.

I'm processing this as I type, feel free to correct:

In theory you create a circuit to respond to a specific frequency range, let's say I want to focus on clean and clear, even, low bass ranges from 20 hz to 100 hz. Narrow for the purpose of how I am thinking through this, I would want a devoted route to pick-up, amplify, blend and balance. Now is that one channel for this example (monophonic) before being mixed and mastered? I then need to understand that current varies based on input volume and frequency, and a driver on the output end attempts to faithfully create wave patterns that match the input.

I know that drivers can respond to certain ranges. In my previous example I need a coil that creates the vibrations needed to reproduce the 20-100 hz range. The incoming signal has to be filtered and amplified in order to faithfully reproduce a very desirable sound.

All things equal, the frequency cutoffs were selected to maximize the ranges to allow an enjoyable product for the consumer, and make the recording and mastering engineer happy.

I have no clue if what I said makes sense, but I feel like it is a solid step one to grasping this theory from my perspective. If so, it makes sense to me as a musician. It would be like wanting to study how best to arrange your performers based on tone quality, blend and balanve during a musical phrase. Then adjust what is needed during the monitoring and eventual mastering phase.

No wonder rare radios/phonographs that could do this would be so expensive. It's akin to building the same equipment today.

Funny how this impacts the harmonic series of any given pitch.

[old_tv_nut]

You mixed a few issues together, I think.
To do stereo, you need two separate channels. So for good stereo, the first thing you need to accomplish is a good single channel; then you can duplicate it to have two.

So then, you need to create a design for a good single channel, which involves what is the frequency response of the transmission channel and amplifiers (which will carry all frequencies of interest), and then the frequency response of the loudspeaker and output amplifier(s). At this final stage, the speaker can be a single speaker, or a 2-way, or a 3-way (or more); e.g., a 3-way with woofer (bass), mid-range "squalker," and tweeter (treble). The final amplifier can be a single one that drives a speaker with one, two, or three parts, or it can be split into 2 or 3 amplifiers, each of which is optimized for the frequency range of the loudspeaker part it is driving.

In systems with a subwoofer for the very lowest frequencies, the subwoofer often is combined with its own built in amplifier and adjustable "crossover" (electronic filter) to select only the frequencies that the subwoofer is good for.

A (monaural) TV set or table radio or even many early console radios would typically have a single amplifier and a single "full range" cone loudspeaker to handle all the sound. It is much more difficult to get the fullest range of sound frequency output from a single speaker, so HiFi speakers typically are three way or two way. Even with 2-way or 3-way speakers, a single wide-range amplifier is common. If separate amplifiers for each frequency band are used for a 3-way speaker, it is referrred to as "tri-amp."

Hope I am understanding your questions, and that this helps clarify.

I think you are interpreting them well.

This clarifies many things I had yet connected; on the other side can this help me teach myself how to do alignments?

If I can apply similar theoretical research; videos, schematics of various circuit types, root mean square, peak to peak voltage (several other concepts in the electronics books I have since collected) and discover other aspects like rf/if, voltage and wave shape etc.?

[old_tv_nut]

To give you an idea, the crossover frequencies for a three-way loudspeaker may be around 500 Hz and 3500 Hz. In other words, the woofer reproduces frequencies from its lowest possible response (maybe 40 Hz or less on a good speaker) up to 500 Hz; the mid range reproduces the range from 500 Hz to 3500 Hz, and the tweeter reproduces frequencies above 3500 Hz to its highest capable frequency (at least 10,000 Hz to qualify as HiFi, and often 20,000 Hz).

Subwoofers may have a crossover around 80 Hz, so they reproduce frequencies of 80 Hz and lower.

Examples of low-fidelity frequency responses:
The typical telephone response used to be about 300 Hz to 3400 Hz, adequate for voice recogniton but lousy for music. AM radio network lines were equalized to carry a wider range, from low bass up to about 5 kHz. AM stations and AM radios also limited the highest frequency to 5 kHz, to prevent interference from adjacent stations on the dial, which otherwise could produce a continuous 10 kHz tone in the sound.

[Electronic M]

There are also many different forms of stereo mic placement, mixing, and listening that achieve different effects of spatial imagery to the listener.

We only have 2 ears but we can detect if a noise in the room with us comes from infront of us or behind us. We perceive this through the differential timing/phase at which the sound reaches each of our ears and the differential timing of echoes off objects around us. In the 50's and 60's there were 'Dynaural' stereo records recorded with 2 mikes in a dummy head designed to pickup the spatiality as a human would hear it, and to only accurately convey that spatial image to someone listening with headphones.

Some stereo is captured with only 2 mikes on several feet apart on stage and designed to give a console listener the effect of sitting in the audience. Some stereo is created by recording each instrument to its own mono tape track then manually mixed to each channel so the engineer/producer can paint their own soundscape by controlling the volume and L/R channel distribution of each instrument on a mixing console....Many pieces of popular music that predated stereo recording were later "rechanneled for stereo" by isolating each instrument (or family of instruments) by frequency using equalizers, then remixing the individual 'isolated' instruments to each channel to paint a soundscape electronically...This process was less accurate than creating the stereo mix live from many individual mikes or tape tracks.

In the 2 mike approach in classical music there are often back echoes off the concert hall that give a sense of spatial depth. Some consoles had built-in reverbs that could add a simulated echo to add synthetic depth. There were also boxes that effectively connected a pair of backchannel speakers in series across only the hot leads of the front channels to place the differential echoes of a concert hall behind the listener.

Another thing to consider is the ability to localize sound and the ways it echoes vary with the frequency of the sound. High frequencies are easy to localize but bass especially DEEP bass is more difficult for human hearing to localize thus modern surround systems usually have 5 tiny speakers that are only good for mid-range and treble, and direct all the bass to a single large subwoofer.

Quadraphonic sound (which died in music but survived in movies thanks to Dolby 5.1) is another fascinating topic to dig into...I could go on quite a bit on that topic as I have collected every record and decoder format and tape format sold in the US.

[old_tv_nut]

Originally Posted by BandDirector

If I can apply similar theoretical research; videos, schematics of various circuit types, root mean square, peak to peak voltage (several other concepts in the electronics books I have since collected) and discover other aspects like rf/if, voltage and wave shape etc.?

The general concepts of frequency and amplitude vs. frequency apply in alignment, but the frequencies you work with there with are much higher than audio frequencies: intermediate frequencies (IF) and radio frequencies (RF). IF and RF are not directly audible frequencies - you need instruments to detect and measure them.

Topics to study:
Amplitude modulation
Superheterodyne receiver

Originally Posted by Electronic M

There are also many different forms of stereo mic placement, mixing, and listening that achieve different effects of spatial imagery to the listener.

We only have 2 ears but we can detect if a noise in the room with us comes from infront of us or behind us. We perceive this through the differential timing/phase at which the sound reaches each our ears and the differential timing of echoes off objects around us. In the 50's and 60's there were 'Dynaural' stereo records recorded with 2 mikes in a dummy head designed to pickup the spatiality as a human would hear it, and to only accurately convey that spatial image to someone listening with headphones.

Some stereo is captured with only 2 mikes on stage and designed to give a console listener the effect of sitting in the audience. Some stereo is created by recording each instrument to its own mono tape track then manually mixed to each channel so the engineer/producer can paint their own soundscape by controlling the volume and L/R channel distribution of each instrument on a mixing console.

In the 2 mike approach in classical music there is often back echoes off the concert hall that give a sense of spatial depth. Some consoles had built-in reverbs that could add a simulated echo to add synthetic depth.

Another thing to consider is the ability to localize sound and the ways it echoes vary with the frequency of the sound. High frequencies are easy to localize but bass especially DEEP bass is more difficult for human hearing to localize thus modern surround systems usually have 5 tiny speakers that are only good for mid-range and treble, and direct all the bass to a single large subwoofer.

I was thinking something along these lines and you put it in words for me. I'm a wet newbie to understanding recording theory, engineering and mastering theory and connecting them to the musical concepts I have learned in my degrees.

There is a section I am teaching focused on Music Production. Since my training is so classically focused, I'm trying to learn the foundational principles on my own. What I would like to do is have a deep enough understanding of each step from content/media creation and mastering, recording live, equipment theory and design, and construction of recording and reproduction devices so that I can develop curriculum that can be interpreted at various musical, or non-musical levels for my students.

There is so much to trim and make age appropriate that I am just beginning to understand (first-year teacher) how easy it is to overteach!

I want to include how to interpret and appreciate the analog world as much as the digital world, of which again I didn't major in media and sound engineering, it is music education! I'm the kind of person that craves the whole experience and understanding of every level of any passion or hobby I may have. Specific to music, fortunately I also teach music theory and orchestra so this is balanced elsewhere.

Originally Posted by old_tv_nut

The general concepts of frequency and amplitude vs. frequency apply in alignment, but the frequencies you work with there with are much higher than audio frequencies: intermediate frequencies (IF) and radio frequencies (RF). IF and RF are not directly audible frequencies - you need instruments to detect and measure them.

Topics to study:
Amplitude modulation
Superheterodyne receiver

The Livingston is a Superheterodyne set, I know that means that the radio generates a competing frequency that cancels out the carrier frequency to allow for amplifying the audible frequency spectrum.

I know that testing equipment requires a scope, specific voltage controlled equipment (variac, though I don't yet posses one), a vttm, bias supply, signal and sweep generation, etc. I'm studying how to use them.

[Electronic M]

Your understanding of superheterodyne operation is wrong I'll explain it at the end.

IF alignment and Acoustics are 2 different concepts that are only LOOSELY correlated.

Ideally with IF alignment, you are looking to get the max gain while trying to keep enough bandwidth for the audio spectrum and (more so on FM) achieve a flat response. IF stages have different audio response curves between AM and FM and Between AM sets designed before and after AM Stereo was established. Theres also different audio curves for AM transmitter Pre-Emphasis depending on if the station is Voice, Mono, Or Stereo music.
Pre-Emphasis and De-Empahsis in radio are Similar conceptually to the RIAA equalization curve for LPs...Basically, if you emphasize (the most noise suceptible) parts of the audio band before transmitting over a noisy medium then de-emphasize them at the receiving end to return audio equalization to normal the De-emphasis reduces noise thus improving the signal to noise ratio (or in LPs reducing record dust/scratch noise).
Alignment can't change a station's preemphasis, It can compensate a bit and at absolute best only try to be a compromise between what is best for each different station. Audio frequency response of the IF is fairly baked into circuit design so trying to adjust an existing reciever for more than an optimization of gain and bandwidth isn't something anyone does (trying would probably drive ya mad).

Osc adjustment is there to try and make the numbers the dial is pointing to accurate.
RF stage adjustment is to optimize sensitivity to weak signals and make sure that sensitivity is even across the band.

Preamble to Superhetrodyne opperation:
First some info on RF and modulation, Radio frequency is a single tone far above human hearing range in consumer sets (though the Navy uses IIRC 75Hz to communicate with submarines in deep sea). We make that tone carry information by varying its amplitude (intensity) or its frequency instep with the changing instantainious value of the information we want to send. AM Modulation makes the curve formed by the peaks and the curve formed by the valleys of the carrier waves match the shape of the modulating audio signal. To recover the audio from an AM carrier you rectify the carrier at the detector which gives you high-frequency DC spikes who's peak values trace the curve of the audio modulation...A filter is used to remove the remaining RF and smooth out the RF rate jaggedness of the audio giving a nearly perfect reproduction of the audio waveform sent. Perfect AM modulation is impossible changing the carrier amplitude at some frequency also changes carrier frequency. It is possible to modulate the peaks of an AM signal with one modulating signal and the valleys with a different modulating signal creating AM double sideband stereo (which was a thing at least while the AM stereo format wars existed in the 80's). Normally both sidebands are modulated the same for mono. Perfect Amplitude Modulation is impossible the carrier frequency changes with the frequency of the modulation. Thus you need a range of frequencies on either side of the carrier to accommodate the modulation frequencies. In mono AM the bandwidth you need is 2X the audio bandwidth you want or are allowed to transmit.

How a Superheterodyne works:
A Superheterodyne is designed to take a limitation of RF amp design and change it into an advantage. Basically an RF amp has a Gain-Bandwidth product...In other words the if you want a ton of gain you have to make the bandwidth it can operate at really small, but if you want a lot of bandwidth you can't have much gain (gain being a measure of the increase in signal an amp produces). So before the Superhetrodyne was invented radios were TRF (tuned radio frequency) basically around 3 Amp tubes that could amplify the entire AM band with adjustable filters between each amp to select the station, the output of the amp chain feeding a detector.
What A Superheterodyne does is convert a selectable RF carrier of variable frequency and convert it to a fixed frequency.
It does this by mixing the carrier frequency with a local oscillator frequency to Produce an Intermediate RF Frequency (IF). Basically when you mix 2 sinewaves (carrier which we will call Fc, and the Oscillator which we will call Fo) we get 4 sinewaves....Fc, Fo, Fc+Fo, and Fc-Fo. Also the Fc+/-Fo outputs retain the modulation that Fc had before the mixing...So you've magically made the carrier a lower and a higher frequency. RF amps usually work better at lower freq so IF is usually chosen to be the Fo-Fc. Basically a superheterodyne varies the oscillator and such that the mathmatical equation Fc(of station)-Fo=IFfrequency is maintained at all times. Theres also a AM preselector filter that tries to reduce all other stations but the one the osc is trying to convert to an IF frequency.
This system results in cheap radios that are extremely sensitive (ie high gain) to weak/distant stations and better at separating adjacent stations on the dial from eachother. The IF amps only need to work over ~1/100 the bandwidth of an RF amp that covers the whole AM Band so theoretically the same tube could make 100x the gain if used as an IF amp instead of an RF amp. Each IF stage usually has 2 tuned filters that reject all frequencies outside the IF band...The more filter stages the stronger the rejection of signals (other stations, atmospheric noise, etc) close to but narrowly outside the band of the desired station....Most radios have 2-3 IF stages plus 1-2 filters in the RF stage and the filtering effect of the local osc frequency conversion which is far more than the 3 filters in an average TRF.

You can demodulate an AM signal by mixing a fixed amplitude RF signal of the E-X-A-C-T same frequency and phase as the station carrier, but that requires PLLs which weren't up to the task in the tube era (unless you wanted your radio to cost as much as your house).

Utterly fascinating to me, this is the math and science I want to learn right now.

[Chip Chester]

To amplify a bit on Electronic M's post --

As EM says, low frequencies are inherently non-directional (hard to locate) and that's due to long wavelengths, mostly. That's one reason why subwoofers are often single-channel, with summed LR content below a certain design frequency. (5.1 and 7.1 theater sound is an implementation of this.) This started because LP records almost always have mono summed low end, due to disk cutting physics. (And most out-of-phase content is summed as well.) During the reign of CDs and now electronic distribution, the mix engineer is freer to experiment with phasing and hard-panned L to R bass... and now that LPs have made a comeback to some extent, the limitation has returned. On LP's, bass instruments can still sound like they come from one side or the other, but that is a byproduct of the higher frequencies of a given instrument -- string plucks, thumps, etc. Down deep in the frequency spectrum, it'll be mono (centered equally in the left and right channels.)

Switching to multi-track music production -- even though a common approach is one track per instrument, it's common to devote a stereo pair of tracks to inherently-stereo instruments: drum set overhead mics, 'fake stereo' Fender Rhodes electric pianos, backup choirs, etc. During the mix, an engineer can vary the spread of these channels all the way out, narrow the image, or move the stereo image slightly left or right as needed to "fit things into the mix".

During one session, I had maxed out my available tracks, leaving just two for the dozen or so percussion toys that needed to be added. Put up a stereo pair of mics, and had them all play at once -- no overdubs for individual tracks, just perform it with the balance and arrangement you want on the final track. Took them just a couple of takes to get it worked out, and saved quite a bit of studio time (and tracks) in the process.

Many books on the physics of music, physics of musical instruments, and recording techniques. The Recording Studio Handbook by John W. Woram (as I recall) was the book to read when I was starting out -- analog multitrack recorder time period. Though some of those techniques are no longer in mainstream use, the basic concepts are still sound. Especially microphone technique. Many more publications/YouTube sources out there, of varying levels of accuracy and expertise. Modern recording usually takes place on laptops in basement nowadays. What used to cost half a million can now be done on a $1K computer using the software that comes with it.

[Titan1a]

One low-frequency experiment I did was to generate a 100 Hz sound and directing it through an open doorway. Almost absolute silence! Unreal!