Frequncy standard use in audio equipment

[JohnnyMalaria]

Human hearing is remarkably forgiving.
For years, movie soundtracks recorded at 24 f.p.s. film rates were replayed on TV at 25 f.p.s., shifting all audio frequencies by around 4.2%.
Did anyone notice?

True for PAL in 50Hz countries. For NTSC in 60Hz, 24 fps films are (were?) broadcast with 3/2 pull-down which doesn't require shifting.

[vk6zgo]

Today's methods of changing speed do it without a pitch shift, so may be less obvious.

It wasn't obvious to most people, just as an SSB received signal can be slightly off "zero beat" without sounding strange.

Strangely, some people seem more sensitive to phase errors between stereo channels, than absolute frequency.

I'm not at all sure how that works, but it seems to be those that listen in very quiet surroundings, or with headphones.

The ABCFM transmitter at Mawson, Western Australia was (I'm not sure, if it is now) fed with the discrete L&R signals, via a combination of microwave bearers & landlines.
(The community FM transmitters that shared the TVW7 site in my later job used composite stereo on a single Programme line.)

Unbeknownst to us at the site, part of the landline section had a failure of one stereo channel, so the phone techs rerouted  that channel only, via another couple of hundred km to restore the service.

One of our techs, who was very much into Classical music, swore he could hear a phase error between the two channels.

We listened on speakers in the quite noisy Tx control room & "Told him he was dreaming."
He persisted, so "just to shut him up" we checked it.

In their wisdom, the EEs at the ABC & Telecom Australia had instituted a pilot system to operate Programme fail alarms.
This pilot was in the form of a low level 15kHz tone on each channel,  R & L.

The alarm box  at the TX end of the programme lines very kindly supplied an output of these tones, so comparing them with an Oscilloscope was easy.

Of course, he was right! the pilot tones were well out of phase.
The phase error on lower frequencies would obviously be less, but to a "golden eared"  person, it was discernible.

On another occasion, at a different place, another audio enthusiast told us that there was intermittent noise on the stereo sound signal from a Commercial TV Tx.

Again, we all listened, hung a 7L12 spec an off the aural exciter, but couldn't either hear it or see it.
At that juncture I had a couple of days off, & returned to find that the guy standing in for me had been called out when the exciter went crazy, spilling white noise right across the sound channels.

Again, the person who was seriously listening in a quiet environment warned about a problem that was not evident to people in a noisier situation, even though they were actively listening for it to happen.

I think the 7L12 wasn't quite up to the job, although I used it later to find interference on a composite FM stereo signal.

[vk6zgo]

Human hearing is remarkably forgiving.
For years, movie soundtracks recorded at 24 f.p.s. film rates were replayed on TV at 25 f.p.s., shifting all audio frequencies by around 4.2%.
Did anyone notice?

True for PAL in 50Hz countries. For NTSC in 60Hz, 24 fps films are (were?) broadcast with 3/2 pull-down which doesn't require shifting.

I avoided saying "PAL", as 50Hz BW systems  were around for years prior to PAL, & the same thing was done with SECAM systems.

One thing that happened everywhere, though, was replaying vintage 16 (& two thirds! ) fps movies at 24 or 25 fps.
The Kaiser's armies didn't look fittingly ominous in history programmes, when they seemed to be prancing along.
Modern systems have restored the gravitas of that formidable war machine.

[Ultrapurple]

I came to this discussion a bit late, but I think a rather obvious point has been overlooked.

Audiophoolery should be checked with a double deaf test...

[coppice]

Double blind testing is very important. However, it takes a lot of testing to reach a meaningful conclusion.

"clearly hear the difference"  should not take that long. If the differences are so minor that it takes forever to detect them reliably, then is the upgrade really worth the money?

Value for money is a different issue. We frequently accept compromises in performance of all sorts of products because of price. The audiophoolery issue is whether there is a perceivable difference at all.

Frequently when you find a performance issue in audio equipment the right recording makes it show up very badly, every time you play that recording. If might be a bass line that hits some horrible resonance, or a transient that provokes some horrible effect. In things like codec research you build a portfolio of recordings which produce horrible noises through existing codecs to see how they perform with the codec you are working on, and some of those issues are far from subtle.

[dmills]

The best codec snafus happen in video however....

There was one of the early motion estimating Snell and Wilcox scalers that in development was notoriously prone to mistake peoples heads for footballs when the input was a game with a panned shot.

Also, grass that looked very, very, strange was a common one on compression codecs.

There is some evidence that close in phase noise on the modulator clocks can be audible, should not really be a surprise, sampling is basically mixing in the multiplication sense so noise sidebands should be expected. 

Jitter was a problem, and it is rather telling that the early protools rigs sounded BETTER with an external clock (Internal XO should always beat a VCO), but that was 25 years ago, today this is something you can fuck up, but you would have to work to get it really wrong.

[bdunham7]

I'm asking because I'm genuinely curious....

I have been seeing reference oscillators of all kinds, including Rb, OCXO, and GPSDO for audo use.  Some report says they can "clearly hear the difference."  I'm thinking for syncing digital recording, etc, regular crystal or maybe TCXO is sufficient and that human ear can't possibly distinguish the difference in phase and frequency stability. 

Is this yet another case of snake oil?  Some of those goes for $4KUS or so.

Just saw this topic, so here's my $.02

Jitter in the digital sample clock usually wouldn't show up as the 'wow and flutter' from the magnetic tape days (although you could perhaps simulate it) so you typically wouldn't hear a pitch or phase difference.  Sample clock jitter simply translates to an amplitude error--in the file if the jitter is during recording, in the audio output if it is in the DAC clock.  Uncorrelated (random) jitter shows up as noise and is less perceptible.  Correlated (periodic) jitter shows up  differently and depends on the audio signal, because that is what it is modulating. 

There have been many arguments on this topic. The audiophool salespeople will claim that picoseconds of jitter are audible unless they are selling you products with 5 orders of magnitude more than that, then they claim it either isn't an issue or that it is the 'right kind' of jitter.  I remember looking at this closely once and concluding that 5ns or less of uncorrelated jitter just couldn't be audible, even theoretically, and people who do experiments generally find much higher thresholds.  Correlated jitter would be harder to make a definitive statement about, but I suspect that this is rarely an issue outside of devices that are broken or badly designed.  Just keep in mind that having a nice clock doesn't necessarily eliminate jitter, it can be reintroduced by power supply noise, interference or other things that may vary the trigger threshold of the clock signal somewhere in the chain.  This is one reason why 'master clocks' and synchronous data in an audio reproduction system are sort of idiotic--you want asynchronous data transfer to a buffer and then one clock right at the DAC.

http://archimago.blogspot.com/2018/08/demo-musings-lets-listen-to-some-jitter.html

Dither, which can be equated to uncorrelated jitter, can be introduced to mask distortions caused by truncation (reducing bit depth) and decimation, but can also be used to lower the apparent noise floor by pushing it up the spectrum.  So on a typical CD, the jitter dither is not uniform, but is distributed so that most of the energy is in the upper octaves where it is less audible.

This stuff is all well known to competent audio engineers for many years now, but not all products take advantage of that knowledge, especially those on the very low and and, ironically, on the very high end.  Of course on the high end all of their silly design mistakes can be compensated for with expensive exotic wood component stands and speaker cables starting at $10,000.

[jonpaul]

Hello all et bonjour a SiliconWizard...

Been reading the thread with great interest.

We were working on digital audio since 1970s (Dolby, THX, Eventide)

Also on  AES Standards commitee  for digital audio transmission and digital microphones 1980s..2017. (AES-42, AES-13 I think...) So I wrote  many AES papers, also SMPTE and NAB.

A few notes please;

 master clocks are usually at studios that create the media. Nowadays a GPS 10 MHz is very accurate, low jitter and low cost eg the Leo Bodnar.

Most consumer devices are playback,  not media creation, so  clock and data jitter is much more important that any clock frequency.

Digital audio transmission (AES/EBU, SP/DIF, AES3id) is on a single serial stream, Manchester encoded with embedded clock. A PLL is ued for clock recovery.

Thus the DAC for reconstruction depends on a recovered clock, from the PLL.

The noise to jitter issue was well researched bak in 1980s, see especially our old friends  Dr Steve HARRIS (Crystal/Cirrus) classis papers (1987?) as well as the fine work of Dr Julian DUNN (RIP)  (Cambridge, AP).

Most of the jitter is due to CM HF asynchronous  noise contamination of the clock and PLL.

We designed special TX and RX ref circuits to reduce the jitter issues, by greatly increasing CMRR and reducing primary to secondary capacitance in the required  transformers.

The fine  results were confirmed by both lab and listening tests.

Beware that any fair listening needs a double blind like the 1970s..1980s ABX relays with random hidden selection.

The Audiophools will object, but the ABX was a wonderful Occams Razor.

The ABX are long gone but easy to build with a flip-flop, oscillator and gold contact relay.


Hope this sparks some interesting response !


Jon

[bdunham7]

Most consumer devices are playback,  not media creation, so  clock and data jitter is much more important that any clock frequency.

You'd be surprised how many consumer devices have ADC inputs.  All of the signal processing, including tone and volume, is often done digitally in a 32-bit processing environment, so any analog line-in, microphone or phono inputs may be converted to PCM.

Digital audio transmission (AES/EBU, SP/DIF, AES3id) is on a single serial stream, Manchester encoded with embedded clock. A PLL is ued for clock recovery.

Thus the DAC for reconstruction depends on a recovered clock, from the PLL.

The noise to jitter issue was well researched bak in 1980s, see especially our old friends  Dr Steve HARRIS (Crystal/Cirrus) classis papers (1987?) as well as the fine work of Dr Julian DUNN (RIP)  (Cambridge, AP).

Most of the jitter is due to CM HF asynchronous  noise contamination of the clock and PLL.

Yes, those synchronous formats use the embedded clock and top-quality systems nowadays use a dual-PLL to recover it--and that is certainly more than good enough provided the signal is good.  Synchronous is needed for video, but playback from music servers nowadays--both local and remote--often uses asynchronous transfer to a buffer and the DAC uses a local clock.  It either works or it doesn't--there's no possibility of degrading the clock with this setup.  On my system, when I stream a radio station, play a FLAC or MP3 file or even play a CD (depending on how I do it) the streams are all buffered and clocked locally--no embedded clock and no PLL.

[MazeFrame]

I'm asking because I'm genuinely curious....

I have been seeing reference oscillators of all kinds, including Rb, OCXO, and GPSDO for audo use.

World Clocks (as used in studios with lots of conversions and digital to digital interconnects) run at 10MHz via 75 Ohm BNC cables.

There are also "re-clockers" that help the less timing stable formats (TOSLINK) keep pace, mostly because some manufacturers cheap out on the inputs.

This may make sense for PA installations on concerts or bigger recording studio setups (= your job depends on the result sounding good). At home, no point to it.

[EPAIII]

TV movies played at 25 fps? Really?

I spent 45+ years in TV engineering. I designed and built TV stations. I installed the film projectors on what is called a "film island" which usually consisted of two 16mm/35mm film projectors, a 35mm slide projector, a TV camera, and the necessary optics and electronics to select one of the projectors to be seen by the camera.

The film projectors used were always run at the correct film fps rate. ALWAYS. Some of these film projectors relied on the AC power for their speed while others used the TV station's synchronizing generator. But in all cases the film was run at the same fps speed that it was recorded with. And that means that the sound track was also being run at that speed, usually 24 fps, not 25 fps. If they were run at 25 fps, the film program or commercial would have ended before it's stated run time and that would have caused problems with air time that would need to be filled. But a 50 minute film did run for a full 50 minutes, not 48 minutes which a 25 fps speed would have produced. Likewise for other run lengths.

There is a fundamental problem with the 24 fps speed of film footage and the 50 or 60 frame rates of the various analog TV standards used around the world. This was solved by mechanical systems that chopped the light path at a rate that was about five times the film pull-down speed. The actual pull-down from one film frame to the next occurred in the black interval between two of these intervals of light passage. A small time lag in the pick-up tubes in the TV camera smoothed the transitions and each TV camera frame received the same number of chopped light in the projector. In NTSC video this process is called a "3:2" pull down. In countries that used PAL or other TV video standards with a 50 frame rate the pull down was "2:2" AND the film being shot for TV was shot at 25 fps, not 24 so again, the speeds matched. This is how the non-compatible difference in TV vs. film frame rates was handled: NOT by running the film projectors at a different frame rate.

I am sure that some 24 fps films were shown in those 50 fps countries but that was not the standard. For the most part, no one noticed a shift in the audio frequencies because there was NONE. And when there was a frame rate difference, I am sure that some, probably most people did notice. 4% is not a trivial difference when sound is involved. It probably drove the music lovers nuts.

More on this:
https://en.wikipedia.org/wiki/Telecine#Frame_rate_differences

Human hearing is remarkably forgiving.
For years, movie soundtracks recorded at 24 f.p.s. film rates were replayed on TV at 25 f.p.s., shifting all audio frequencies by around 4.2%.
Did anyone notice?

[bdunham7]

TV movies played at 25 fps? Really?

Yes, in markets where the screen vertical scan rate was 50Hz.....

b/t/w I've actually seen video that has been processed from 24FPS to 25FPS by simply repeating every 24th frame, a 24:25 pull-up if you will.  It looks hilarious.  But the soundtrack was OK.

"Film shot for TV"--yeah, OK.  But what if it wasn't? 

[vk6zgo]

TV movies played at 25 fps? Really?

I spent 45+ years in TV engineering. I designed and built TV stations. I installed the film projectors on what is called a "film island" which usually consisted of two 16mm/35mm film projectors, a 35mm slide projector, a TV camera, and the necessary optics and electronics to select one of the projectors to be seen by the camera.

The film projectors used were always run at the correct film fps rate. ALWAYS. Some of these film projectors relied on the AC power for their speed while others used the TV station's synchronizing generator. But in all cases the film was run at the same fps speed that it was recorded with. And that means that the sound track was also being run at that speed, usually 24 fps, not 25 fps. If they were run at 25 fps, the film program or commercial would have ended before it's stated run time and that would have caused problems with air time that would need to be filled. But a 50 minute film did run for a full 50 minutes, not 48 minutes which a 25 fps speed would have produced. Likewise for other run lengths.

There is a fundamental problem with the 24 fps speed of film footage and the 50 or 60 frame rates of the various analog TV standards used around the world. This was solved by mechanical systems that chopped the light path at a rate that was about five times the film pull-down speed. The actual pull-down from one film frame to the next occurred in the black interval between two of these intervals of light passage. A small time lag in the pick-up tubes in the TV camera smoothed the transitions and each TV camera frame received the same number of chopped light in the projector. In NTSC video this process is called a "3:2" pull down. In countries that used PAL or other TV video standards with a 50 frame rate the pull down was "2:2" AND the film being shot for TV was shot at 25 fps, not 24 so again, the speeds matched. This is how the non-compatible difference in TV vs. film frame rates was handled: NOT by running the film projectors at a different frame rate.

I am sure that some 24 fps films were shown in those 50 fps countries but that was not the standard. For the most part, no one noticed a shift in the audio frequencies because there was NONE. And when there was a frame rate difference, I am sure that some, probably most people did notice. 4% is not a trivial difference when sound is involved. It probably drove the music lovers nuts.

More on this:
https://en.wikipedia.org/wiki/Telecine#Frame_rate_differences

Human hearing is remarkably forgiving.
For years, movie soundtracks recorded at 24 f.p.s. film rates were replayed on TV at 25 f.p.s., shifting all audio frequencies by around 4.2%.
Did anyone notice?

That may have been your experience, but it was not mine, having spent a similar time working in the TV industry.

In 50Hz countries, it was common to run ordinary film at 25fps rate.
When we were taught about telecine equipment back in the 1960s, a big thing was made about how North America, with their 30fps system, had to use complicated "pull down" systems to present film run at 24fps.

It was pointed out how fortunate 50 Hz countries were in that regard, due to the quite small difference between the standard film rate & the fps in 50 Hz countries.

Certainly in Oz, most of the telecine chains in use from the 1960s to the early 1980s were simple "projector looking at a vidicon/plumbicon camera" assemblies.
There were a few legacy "flying spot scanners", but they were relatively rare.

During the early to mid '80s, "line stores" came into use in telecines.
This allowed line readouts at a different rate to that which the projector was operating.
These were in time, supplanted by frame stores, allowing the film to be played at any rate, & seamlessly converted.

Your comment about "run times" is a bit of a "red herring".

Films shown on Oz Commercial TV back in the pre Videotape days were pretty savagely "hacked about" to fit the number of ads required, so any run time difference could have easily been absorbed.
Each film reel was, after the insertion of ads, marked with its run time, so the operators knew exactly how long each ran.

Despite that, films did run short or long----- they still do, even with all the modern stuff, as do theatre movie presentations from time to time.


The ABC (non- commercial) filled spaces between programmes with short features or musical interludes, which could be, & were cut short as required.

P.S. this is a quote from your linked site:-

"However, during the analogue broadcasting period, the 24 frame per second film was shown at a slightly faster 25 frames per second rate, to match the PAL video signal. This resulted in a fractionally higher audio soundtrack, and resulted in feature films having a slightly shorter duration, by being shown 1 frame per second faster."

[Ultrapurple]

"I'll Be Back.
_{(A little sooner in 50Hz countries.)}"

[coppice]

i recently got into reading up and playing with some "CHi-Fi" stuff (for free)

1 of the manufacturers from china is the firm called "SINGXER", the equipment is SU-2 and SU-6. part of the marketing tagline is "femtosecond clock" (mmmm?)

the actual clock in use inside have jitter in under 1 pico-sec
already, it is creating a circle of hype where people claim to be hear enhancements in sound

however, in SU-2, it claims it is able to inject GPSDO 10Mhz clock to "do something" ... i dont quite understand it.
there is very sparse information about what improvement it can do, and by what means it stabilizes the internal crystals for 44.1/48k.

the 2 crystals in use is not OCXO. but many thinks it is working in that way, misled by some discussion in some other forum. CCHD-957 for SU-6 and AS381B for SU-2. i browsed many similar pdf, but they are TXCO. did i read wrong?

i think this is a case of a "USB memory stick with no chips inside", for SINGXER.

The use of GPS can be practical, or it can be BS. The legitimate goal is for mixing from multiple sources. If the sources are all locked to a super accurate clock they can be mixed freely without sample slips. The BS use is to claim a super accurate sample rate matters for listening purposes.