How to pack a stereo signal in one record groove(vinylrecorder.com)
vinylrecorder.com
How to pack a stereo signal in one record groove
http://www.vinylrecorder.com/stereo.html
5 comments
If you listen carefully to a vinyl record playback you can hear, very faintly, the start of a track at little before it actually starts. A 'pre-echo' that starts exactly one turn of the table before.
The master record is cut in a relatively soft material. Copper, I think. The force of cutting removes material from the groove but also compresses the remaining material somewhat. This distortion may spill over into the neighbor groove just cut one turn earlier. The neighbor groove is silent between tracks. So in this silence you may be able to hear a faint copy of the modulation one turn later. This ghost signal gets more volume the tighter you pack the groove spiral, which you may want to do tighter to obtain longer playing time.
Interesting! I wonder if this has been deliberately exploited by dance producers...at 45 rpm and a bpm of 135, this will yield a 3-beat delay which sits nicely with a 4-beat base rhythm. Dance music production techniques owe a lot to reggae and dub, and as record producers working with low-quality equipment in Jamaica found ways to creatively abuse their gear.
For example, there's a trick whereby you can turn a regular mixing desk into a low or high pass resonant filter by feeding the same signal to two channels and inverting one (so they cancel out), and then EQing the other channel aggressively. So if you can't beat the 'echo' resulting from the manufacture, you might as well look for ways to incorporate it.
For example, there's a trick whereby you can turn a regular mixing desk into a low or high pass resonant filter by feeding the same signal to two channels and inverting one (so they cancel out), and then EQing the other channel aggressively. So if you can't beat the 'echo' resulting from the manufacture, you might as well look for ways to incorporate it.
Another reason for turning it by 45 degrees is backward compatibility. If you play a stereo disc on a mono player you want to get a "mix" of the channels instead of one of the channels.
that doesn't seem like it would work.
In a stereo recording, the two channels are arranged to drive the record cutter head at an angle of 45° to the vertical, effectively encoding each channel in the left and right V-shaped walls of the record groove. This system worked well, since it provided full compatibility with a monaural pick-up, so stereo records could be played on older mono equipment.
From: http://en.wikipedia.org/wiki/Magnetic_cartridge#Stereo_repro...
See drawings of pick-ups above on that page. The vertical movement of a monaural pick-up stylus will be the sum of the two stereo movements. Each channel contributing with an sqrt(2)/2 amplitude of its 45° amplitude.
From: http://en.wikipedia.org/wiki/Magnetic_cartridge#Stereo_repro...
See drawings of pick-ups above on that page. The vertical movement of a monaural pick-up stylus will be the sum of the two stereo movements. Each channel contributing with an sqrt(2)/2 amplitude of its 45° amplitude.
thanks, I was trying to work the math in my head and was making it WAY harder than it needed to be.
This is why, on most vinyl albums, the low end/bass sounds are panned dead center.
Lots of low end panned hard left or hard right can result in the needle jumping out of the groove, and can result in unnecessary compromises in mastering.... there's one album in particular that doesn't have hardly anything above 5k on it, and it's because they decided to pan the bass hard left.
Lots of low end panned hard left or hard right can result in the needle jumping out of the groove, and can result in unnecessary compromises in mastering.... there's one album in particular that doesn't have hardly anything above 5k on it, and it's because they decided to pan the bass hard left.
that's very interesting. I had learned that no one bothers to pan bass frequencies because IRL they are difficult to localize. For example, no one at a live show could tell you if the bass rig is on stage right or left. So, a recording with hard panned bass would just sound wrong.
For live sound this is mostly true, but for recorded music there can be artistic reasons to pan low frequencies to one side or another. (Just like there could be artistic reasons to do just about anything.) It seems to be more common in electronic music rather than rock, and there certainly are jazz records where the upright bass isn't dead in the middle.
I always thought that's why most systems have a single subwoofer. What's the album with the hard panned bass? I'd love to hear it (or know why they chose to do that)
Electroputas "3" is the record I was thinking of. There's basically no high end on it. It was so noticeable that I actually emailed the label about it, thinking I got a bad press or something, but to their credit they admitted that the bass panning made it impossible to master the vinyl cleanly. (Electroputas are a really good band, by the way. Not sure if they're still around.)
Stories like this abound. The more bass you want on a record, the wider the grooves need to be, which gives you less time per side; the more playing time you want on each side, the narrower the grooves need to be, and you don't get as much dynamic range, loudness, or frequency bandwidth.
Another example: there's some Yo La Tengo record, I forget the name (it's been a while), but it's a one sided 12". The running time on that one side was extreme for a rock record - something like 28 minutes or so - and as a result, it sounds terrible.
Stories like this abound. The more bass you want on a record, the wider the grooves need to be, which gives you less time per side; the more playing time you want on each side, the narrower the grooves need to be, and you don't get as much dynamic range, loudness, or frequency bandwidth.
Another example: there's some Yo La Tengo record, I forget the name (it's been a while), but it's a one sided 12". The running time on that one side was extreme for a rock record - something like 28 minutes or so - and as a result, it sounds terrible.
Super cool!
But I didn't understand the last panel ("Maximum Stereo or Mono and 1 channel 180° phased")... and, also, how did they do quadraphonic?
But I didn't understand the last panel ("Maximum Stereo or Mono and 1 channel 180° phased")... and, also, how did they do quadraphonic?
Inverting the phase of a mono signal and feeding it to one side of a stereo channel (with the original on the other) isn't true stereo, but flipping the phase of one channel fools the brain into believing it has a spatial dimension - the sound seems louder and more vibrant although in reality all that's happening is that the left and right channels are interfering with each other.
It was a trick used on a lot of old record when the music industry was moving from mono to stereo recording, but still effective if you're stuck with a mono sound source (quite often the case in film and TV) and you need to beef it up without changing the timbre of the sound substantially.
It was a trick used on a lot of old record when the music industry was moving from mono to stereo recording, but still effective if you're stuck with a mono sound source (quite often the case in film and TV) and you need to beef it up without changing the timbre of the sound substantially.
Wait, isn't that the same effect as a mono signal with one speaker wired backwards? If so, that 'head-swimming' sound that results is awful, and the out of phase drivers will kill much of the bass response.
I'm very surprised that this trick was ever used in industry; after all, if someone listened to that old film on a single speaker TV, the signals would sum to zero and you'd have no audio at all! (And yes, our local cable company has managed this very stunt from time to time...flip to mono and the audio goes away.)
I'm very surprised that this trick was ever used in industry; after all, if someone listened to that old film on a single speaker TV, the signals would sum to zero and you'd have no audio at all! (And yes, our local cable company has managed this very stunt from time to time...flip to mono and the audio goes away.)
Yeah, it can go wrong and it does sound pretty awful. It's good practice to check the final mix out in mono for just that reason but stereo is ubiquitous now and people often don't bother any more.
Quadraphonic has 2 more channel 'encoded' in a high frequency carrier (50 kHz maybe) which can't be heard on regular mono or stereo equipment. So each of the 2 stereo channels will have this carrier signal encoding one additional sound channel.
In the next episode - "How to use sparks to dial a phone number" :)
I used to have a party trick whereby I'd dial a number by tapping the receiver of a phone on a table to emulate pulse dialing. It probably still works on landlines, but it's so long since I had one that I don't know if they support pulse any more.
That actually worked? I thought you had to tap the line switch (the switch the receiver pressed down on when not in use), not the receiver. Which would make sense, to reduce interference from sound coming through the receiver while dialling.
Supposedly pulse dialling still works on (most?) new telephone equipment. That's what the vendors of classic phones I've been looking at are saying, anyway.
Supposedly pulse dialling still works on (most?) new telephone equipment. That's what the vendors of classic phones I've been looking at are saying, anyway.
It does (or at least did) - try it if you know someone with a landline. I was surprised too, as I just tried it on a whim one day. I grew up with pulse telephones and particularly loved the chunky sounds of old models so I used to play with the phone a lot as a kid.
You had to have a good feel for the timing of the pulses or the trick didn't work. Taps and clicks broadband noise with a very sharp transient so it would have to be a very noisy environment indeed for the ambient noise to interfere. I've never played with any analog exchange equipment but the electronics to detect and count pulse dialing doesn't need to be very sophisticated.
You had to have a good feel for the timing of the pulses or the trick didn't work. Taps and clicks broadband noise with a very sharp transient so it would have to be a very noisy environment indeed for the ambient noise to interfere. I've never played with any analog exchange equipment but the electronics to detect and count pulse dialing doesn't need to be very sophisticated.
I once heard that years ago at MIT, they used to vandalize the physical switch on campus by dialing a number, then rigging the handset mic wires to the wall outlet. Since there was an actual physical connection of conductors from one handset to another, the increased current would sometimes somehow stick all of the relays in place, and the phone on the end would keep on ringing loudly until something burnt out.
But the electronics would be different if you did it by tapping the line switch, than if you did it by generating a sharp noise on the line. After all, pulse dialing is also called 'loop disconnect dialing':
http://en.wikipedia.org/wiki/Pulse_dialing
"The pulses are generated through the making and breaking of the telephone connection (akin to flicking a light switch on and off); the audible clicks are a side effect of this."
It's confusing.
"The pulses are generated through the making and breaking of the telephone connection (akin to flicking a light switch on and off); the audible clicks are a side effect of this."
It's confusing.
Well, if you think of it in terms of voltage rather than switching, the audio transient will cause similar spikes in the voltage amplitude. I think that the analog circuits in exchange hardware are only interested in the shape rather than the origin of the pulse but as I said I've never looked at such equipment directly.
I have a modular synth here and have managed to rig up a patch that counts audio pulses (incoming signal goes through a rectifier and an envelope follower and on to an accumulator) and generates an appropriate tone for the counted number after a ~1/4 second pause, and doesn't much care whether I'm opening and closing the circuit or just tapping the front of the microphone. It's not a very complex circuit so maybe analog exchange switching works on a similar principle.
I have a modular synth here and have managed to rig up a patch that counts audio pulses (incoming signal goes through a rectifier and an envelope follower and on to an accumulator) and generates an appropriate tone for the counted number after a ~1/4 second pause, and doesn't much care whether I'm opening and closing the circuit or just tapping the front of the microphone. It's not a very complex circuit so maybe analog exchange switching works on a similar principle.
It still works on Norwegian landlines.