Topic: Is levels peaking at 0 to high?

[firemt66]

Hello,
            Recently I recorded a few shows on my Samsung phone using the internal Mic, they sound pretty good.the levels peak at 0, At least in audacity that player only goes to 0 so i cant see if it goes higher, I hear no distortion when I listen back to them. When I use my M10 usually it peaks around negative 5 or 4. Maybe sometimes even lower just so I don't have to keep checking the levels. Is there a player that goes above 0 so I Could See if the levels are actually Higher? I bought one of those usb cables that connects to my phone so i can use my CA 14 card mics and it worked fine, again i heard no distortion with playback. Thanks for any input...Scott

[firemt66]

I've just discovered an app for androids ASR Voice Recorder and it has  gain control. I lowered it to -3 recorded something at home here and the levels peaked at -3 in wav 1411 kbps 48mhz 16bit.there is no option for 24bit.The app has an option for 32 bit wav. Should I go with that or just the regular wav option?

[goodcooker]

If it sounds fine it is fine. For a recording made with the internal mic of a phone I wouldn't sweat it too much if the peaks that hit zero didn't cause distortion.

For phone recordings with CA mics and no preamp I'd just go with 16bit/48kHz. I don't think there's any real gains to be had with that setup recording live events at higher resolution.

[adrianf74]

If it sounds fine it is fine. For a recording made with the internal mic of a phone I wouldn't sweat it too much if the peaks that hit zero didn't cause distortion.

For phone recordings with CA mics and no preamp I'd just go with 16bit/48kHz. I don't think there's any real gains to be had with that setup recording live events at higher resolution.

Pretty much it -- if it's not sounding clipped, it likely isn't (terribly) affected.  You could always run the file through de-clip in iZotope RX Audio Editor to see if there's any real issue.  Other than that, try and set levels a little lower if you can to avoid this in the future.

[firemt66]

If it sounds fine it is fine. For a recording made with the internal mic of a phone I wouldn't sweat it too much if the peaks that hit zero didn't cause distortion.

For phone recordings with CA mics and no preamp I'd just go with 16bit/48kHz. I don't think there's any real gains to be had with that setup recording live events at higher resolution.

Pretty much it -- if it's not sounding clipped, it likely isn't (terribly) affected.  You could always run the file through de-clip in iZotope RX Audio Editor to see if there's any real issue.  Other than that, try and set levels a little lower if you can to avoid this in the future.


Unfortunately I don't have izope to try that and phone apps that i have has no recording level control other than not higher than 0 apparently. However I do have auducity. this new app i d/l this morning does have gain control for which i lowered to -6. Guess I'll find out this thursday when I go to my next show to see actually where the recording level peaks at...

[kuba e]

This is a logarithmic scale. The highest signal is 0 db, the lowest signal is -96 db (for 16 bit, -144 db for 24 bit). In Audacity, you can also switch to a linear scale (1 - 0 - -1).

When you record too much amplified audio signal, the signal will be clipped. The signal does not fit in the given range. What exceeds 0 db will be cut. If only a small part of the signal is cut off, it is possible to tolerate it. But when the larger peaks of the signal are cut off, it sounds very disturbing . And it can no longer be corrected afterwards. When you turn down the volume in Audacity, the clipping stays there, it just lowers the overall volume. There is a special iZotope De clip tool mentioned by Adrianf74. It uses an algorithm that artificially restores cut peaks. But that's just a rescue, it's the best to record without clipping.

When recording, you need to amplify the signal so that it does not exceed 0 db. Try to find an app that allows gain control. There is also a lower limit. The signal should not be too low, because then the noise increases (noise/signal ratio). What level to set is a question that has been discussed a lot here. I usually set the level to -9 db. It gives me some headroom for loud sounds and the noise/signal ratio is good.

The app has an option for 32 bit wav. Should I go with that or just the regular wav option?

Go with 16 bit. Self phones don't have 32 bit A/D converter.

[jielkade]

Try the german app "field recorder".

[bonghitwillie]

i always try and keep it well below 0 bc you cant fix if over recorded.

[morst]

Digital recording only goes up to -0dB as "full scale" for samples.
The interpolated waveform can peak over -0 dB, but samples in the PCM world can't be higher than that, mathematically. Different software and hardware handle this in various ways, so some might be able to meter above full-scale, but I can't think of an example. Some peak meters or limiters only light up when there is more than one sample in a row at full-scale.

For practical purposes, if only the very peaks of the waveform hit full-scale for one sample at a time, you should be in great shape.
Peaking on the snare drum or other short time duration sound will probably not be audible the same way that peaking on a continuous tone would be, such as a loud sustained note on a bass guitar...

I hope this helps.

[DSatz]

It's easy to assume that all metering nowadays works the way we want it to, because there's really no good excuse for it not to. However, there could be exceptions including well-intended ones (e.g. if the designer thinks, "no one can possibly hear a single-sample overload"). So it's worth loading the recorded waveforms into an editing program on a desktop computer and looking at the peaks in some detail. See whether they appear to run into some kind of limiting or not, or whether they seem unconstrained.

In live recording it is quite rare for signal peaks to remain uniform for any substantial length of time; when they do, it usually indicates some kind of electronic processing (limiting) along the way that is forcing them to fit a common peak amplitude. Whether that is occurring in your equipment or in some preceding equipment in the venue, you'll need to determine case by case. Some recorders and ADCs do, indeed, smash samples that come close to 0 dBFS--and sometimes you want that, at least for safety. But in live recording generally it's better to record first and process later, rather than to "bake" a certain kind and amount of processing into the only recording that you have.

With 16-bit recording, it's very rare that a recording venue is so quiet that you have to run very close to the limit. Typical real-world occurrences make that a dangerous practice; even if you get a run-through of a piece, the performance may easily be 2 - 3 dB louder than the rehearsal at some point or other. After years of analog live recording on open-reel tape, where the dynamic range was distinctly less than 16-bit digital and I got to be a bit of a hotshot at setting my levels so that they just went into the red briefly on peaks and came right back out again, the idea of crowding up against the absolute limit of a digital recording in a live situation makes no sense. If your peaks are anywhere in the top 6 dB without going over, count that as a complete success.

Again, I mean that advice for live recording. In dubbing or processing situations, with the live recording safely "in the can", then it does make sense to approach 0 dB closely IF you're sure of your metering. But that's a real IF. The PCM-F1, for example, had digital meters that were driven from a rectified, analog signal at the input, and they could under-read by as much as 4 - 5 dB in live recording situations; only the "PEAK" indicators were digitally driven, and even those didn't come on unless there were three or more consecutive overloaded samples. It was, after all, a consumer device even though pitched to advanced consumers. Many DAT recorders had similar metering.

--best regards

[morst]

Hey DSatz - if you publish a book, I'd like to buy it, or if you are considering it but need an editor, I'd be up to help proofread/edit.


I always appreciate your insight.
 

[rocksuitcase]

Hey DSatz - if you publish a book, I'd like to buy it, or if you are considering it but need an editor, I'd be up to help proofread/edit.


I always appreciate your insight.
 

seconded. I knew Ron Streicher. I presented at AES in 1986/1987. Dsatz, YOU have a great method for explaining this type of knowledge. It is one thing to know acoustics or microphonics, another to explain it in terms most can understand.
I think my first mic textbook at university was written by a John Eargle?   https://www.abebooks.com/book-search/title/eargles-microphone-book-mono-stereo/author/john-eargle/

[fireonshakedwnstreet]

I want to have at least 10db of digital headroom no matter what happens. You can compress/limit/normalize in post and make it as loud as you want.

[DSatz]

fireonshakedwnstreet, just for me, could you please tell me whether you mean:

(a) you want to set your levels so that 10 dB goes entirely unused permanently at the top (i.e. for you, a recording with its highest peak at -10.1 dB is a more successful recording than one with its highest peak at, say, -4 dB),

or do you mean:

(b) you set your levels ~10 dB lower than what you think is probably necessary, to allow for unexpected occurrences--and then the more of that 10 dB that actually gets used, the more that proves the value of your strategy.

Is it your actual goal to obtain a live recording with 10 dB of unused dynamic range at the top, in other words?
Or is your approach a means to a different end, i.e. being very, very sure not to over-record?

I'm asking because I think that different people think of "headroom" in these two rather different ways. Or maybe you mean something else by it that I don't know about. (Hint: In professional audio the meaning of "headroom" is definitely (b), while anything like (a) is generally avoided except as a last resort, e.g. to work around the limitations of some interface or piece of equipment that has an overload problem.)

--best regards

p.s. Many thanks for the kind words in the messages above.

[SMsound]

If you use a recorder with limiters, then the limiters will kick in somewhere in the region of -7dB. Even with good analog limiters like the ones in my MixPre, I can hear the difference between limiters and no limiters (even after loudness matching). It's a subtle difference, but I definitely prefer the no-limiter sound.

These days, with low-noise preamps available in prosumer grade recorders, why would you come close to -0... I record classical and opera in quiet environs and *really* care about minimizing noise, but I shoot for peaks at -18dB... the ambient noise in an empty concert hall is still higher than MixPre's native noise at -18dB, so there is zero value in pushing it higher and taking risks*

*technically, maybe there's value from a signal-to-noise maximizing perspective? I don't know. Still see no point to doing this in practice.

[admkrk]

I have the limiters turned off in all my recorders, or at least I think they are. Some bands play at a more or less steady sound level and having the recording peaking around 0 is fine. Some bands on the other hand can be averaging around -20 and still end up full of clips. That is the bit that frustrates me.

[fireonshakedwnstreet]

DSatz, yes it would be the latter. Average level would be around -16dbfs with peaks around -6.

[DSatz]

fireonshakedwnstreet and SMsound, thanks for the replies.

I've been quite puzzled about where the other interpretation of "headroom" came from, where people think that it's a more or less wide zone below 0 dBFS that should be kept clear altogether. I think that it might come from the very dishonest polemics against digital audio that many audiophile magazines waged in the 1980s; they used to debunk (or so they pretended) 16-bit PCM's claim to have greater dynamic range than analog recording, in part by subtracting 10, 12 or more dB from the top, claiming that this was the "headroom region" and shouldn't be counted. (Of course, though, they counted every last dB of headroom for analog recording, despite the sharply increased distortion that occurs in that region.)

SMsound, there's no one norm for limiter behavior, and a circuit that starts to restrain the recorded levels as low as -7 dB is a bit of a "soft knee" in my book, between being a limiter and a compressor. I'm more used to "harder" limiters that kick in just below the limit of the medium, and that hold the line there. In any case I agree that anything more than the smallest amount of limiting will change the character of sound, and I don't want any non-emergency limiting in any live recordings that I make. It's better to under-record by 6 - 8 dB than to over-record by 3 or 4, especially if the system doesn't overload gracefully (some do and some don't). Also, with stereo recording it's very important that any limiters be "ganged" between channels. The limiting action, if it occurs, shouldn't cause the stereo image to shift left or right; that's more distracting than momentary "clean" overloads.

--Classical and opera are what I mainly record as well, but even with a 24-bit recorder and quiet preamps, I'm pretty sure that -18 dBFS for one's "all-evening" peaks is lower than optimal, especially if your recordings are ever played back in analog.

But circumstances do vary. It's impossible to calculate accurately with so many unknowns. Preamp noise depends greatly on gain settings, for example. The published noise specs for nearly all preamps are obtained at their maximum gain settings (50 - 60 dB or even greater), where the equivalent input noise is nearly always at its lowest by a significant amount. In concert recording I usually run preamps at ca. 30 - 35 dB gain. Around 20 years ago I measured the noise of all my preamps at ca. 30 dB gain; their relative rankings came out very, very differently from full gain. One of the quietest preamps at 30 dB gain, for example, was the noisiest one at full gain by a considerable margin.

And the noise spectra of condenser microphones vs. preamps are typically quite different, as is the prevalence of "shot" (impulse) noise vs. smooth, continuous-sounding noise. All in all the total, effective, potentially audible noise that you'd get from any given combination of microphone + preamp ... as I said, it's tricky, and back-of-the-envelope arithmetic isn't up to the task. So I stick with the accustomed practice of maximizing the signal levels at each step along the way while taking care to avoid overload. Nothing fancy, just basic principles.

-18 dB, incidentally, seems to come from Sony's old recommended practice for the PCM-1600, -1610 and -1630, which was to set 0 VU = -18 dBFS on steady tone. (I worked quite intensively with those processors in various East Coast recording and LP mastering studios back in the day.) But that recommendation was based on realizing that VU meters are "syllabic" rather than peak-responding, and that people quite properly let them go up to +3 VU for momentary signal peaks. The result with actual program material is that signal peaks on the digital side would commonly reach about -6 to -3 dBFS. Back then, dither wasn't widely understood and with many recorders (including the professional Sony processors that everyone used for CD mastering, in their default settings), distortion and "granular noise" increased quite audibly at lower signal levels. There was never any intention of leaving a wide zone unused near the top of the range!

All told, to me it seems very likely that there would be some s/n benefit if you raised your levels at least somewhat, especially if you consider the noise of your recorder's (or any media player's) analog output circuitry.

--best regards

[SMsound]

David,
I always learn something from your posts.

Something related I have been wondering about: Schoeps recently released some videos that state that with their mics, you maximize signal-to-noise ratio right around 20dB of gain. How should we think about reconciling the Schoeps-optimal 20dB of gain with the recorder-and-source optimal amount of gain that puts you closer to 0dB? (can't seem to find the video, but maybe this one: https://www.youtube.com/watch?v=aSAZN-B9DpU&t=2535s)

Re: limiter threshold, I don't see the limiter threshold on my MixPre published online (I can probably find it in settings somewhere), but the Sound Devices 6-series literature states that the limiter kicks in 'a few dB shy of clipping'. I have observed that the orange limiter lights go on when the meters say I have more than a few dB of headroom...maybe there were a couple samples that went loud and didn't move the meters but alerted the limiter.

I have also noticed that aiming for -18 dB on a Centrance MixerFace resulted in unusably noisy recordings, but on my MixPre-6 (24 bit) I have ambient noise floor above MixPre noise floor when I aim for -18, and this leaves maybe (?) 15dB of headroom in case the soprano gets excited (happens a lot). In practice, I should clarify that I usually ask the soprano to sing me one of the forte passages and put that at -18, but they often don't sing the (louder) cadenza during setup (you can only sing so many cadenzas per day...)...the result is probably more like a -12dB peak level, but just for a moment at the end of the song, and the rest at -18. I did have a show over the summer where I got more aggressive and aimed for -12dB during setup, and a soprano banged the limiters on her last note.

Your 'linked channel limiter' idea makes a lot of sense and I will do that from now on.

[DSatz]

I really wish that you could find the specific statement. Hannes is a down-to-earth guy and totally knows what he's talking about--incidentally he's an active performing musician in his off hours, and a very good one, I think (see https://www.youtube.com/user/nicaroschea). But the statement the way you paraphrased it, in and of itself, honestly doesn't make sense to me.

The preamp gain setting that gives the best s/n ratio depends on several variables, not the least of which is the peak amplitude of the program material that you're recording at the location where your mikes are. Different microphones have different sensitivities and equivalent noise levels. And as I said, different preamps have distinctly different equivalent input noise levels at different gain levels, and at one gain level preamp "A" may be quieter than preamp "B" while at a different gain level the reverse may very well be true--and even that will depend on the source impedance of the microphone (in the tests that I spoke of, I was using a test head connected to a Schoeps CMC microphone amplifier, with the preamp's phantom powering turned on). There's also the input noise and overload point of your recorder's (or a/d converter's) analog input electronics to consider.

So I don't see how there's any way that any one setting, such as 20 dB, can be considered optimal even generally. For my typical setups, that's definitely not enough gain.

--best regards

[SMsound]

I really wish that you could find the specific statement.
So I don't see how there's any way that any one setting, such as 20 dB, can be considered optimal even generally. For my typical setups, that's definitely not enough gain.

I think this is the video/statement I am remembering:
https://youtu.be/2st7KzoEHlo?si=Adw6ObxceFTEvOED&t=1729

They show a plot around 28:50 of dynamic range vs. gain where dynamic range peaks around 15-20dB of gain.

[Gutbucket]

Following along. Mr Satz, can you clarify a couple lines in your post above for me?  You said-

"..especially if your recordings are ever played back in analog." [and later..] "All told, to me it seems very likely that there would be some s/n benefit if you raised your levels at least somewhat, especially if you consider the noise of your recorder's (or any media player's) analog output circuitry."

Its the statements after "especially" that caught my eye.  Does that assume analog transfer of the raw recording from the recorder? Or that a digitally transferred recording will not have levels adjusted in the process of producing the released recording?  If so, I see how what you describe can be the case.  However, most tapers are digitally transferring the raw recording to computer and adjusting levels in editing software as part of the process of producing the finished output they and others will listen to.  That process typically includes increasing signal level to remove excess recording headroom that no longer serves a useful purpose in the released version, and instead becomes a potential detriment for the reason you describe.  When that is the case, I fail to see how analog playback or analog output circuitry noise would effect a recording originally made with excess headroom differently than one made with optimal recording levels.  I understand how the first may potentially have a lower s/n due to a higher noise floor, but that would be baked into the recording and independent of playback.

Just making sure I'm understanding you correctly and not missing something there.

Thanks for the discussion - in particular your mention of the different noise spectra of condenser microphones in comparison to preamps and the noise characteristics of preamps at various gain settings.

[DSatz]

SMsound, the topic from about 24:00 to about 30:00 in the recorded talk is dynamic range, which takes into account both the noise floor and the maximum SPL of the signal chain. He's very concerned to compare apples with apples: the digital microphone with the analog microphone PLUS the preamp that you have to use with it.

As he says (ca. 26:30), the CMD 42 isn't limited by having to pump out high voltages (more electrical power) at higher SPLs, so its maximum SPL is a few dB higher than that of the CMC 6*. But when you bring an analog mike preamp into the picture (ca. 27:15), the clipping point of its output becomes a much more serious limiting factor when you get near the maximum SPL of the microphone. In the slide at the point that you mentioned, he's saying (around 28:20) that you would need to limit the gain on your preamp to about 15 - 20 dB in order to avoid this overload at the maximum SPL.

He's definitely not saying that this is an optimal setting--on the contrary! His whole point is that analog preamps typically have lower input noise than the microphone's output noise only when set to higher gain levels than that, such as 30 - 35 dB. (This was shown in earlier slides, ca. 25:15 - 26:00.) Thus there exists no one gain setting for a typical analog preamp that both accommodates the maximum SPL of the microphone without clipping, and simultaneously offers the lowest noise for the quietest sounds that the microphone can pick up. If you need to record both at the very highest and the very lowest SPLs without touching any gain settings in between, the digital microphone offers a definite advantage as compared with the analog microphone--given that typical analog mike preamps can't put out 10, 20, 30 or more Volts (not that you really would want such levels to occur in practice).

But then he goes on to say two things. (ca. 28:55) "With suitable operation of the preamp, I can obtain performance from the analog system that is exactly as good, or nearly as good, as that of the digital microphone. But I must operate it in a suitable way; I must really set exactly the gain level that suits my application exactly." Secondly, (ca. 29:50) combination analog mike preamps and A/D converters exist that use multiple analog gain stages and gain ranging--and he says that such equipment can produce results that are very nearly equivalent to digital microphones.

--best regards

* note from me: This assumes a standard 1 kOhm load on the CMC 6. It can put out higher voltages if the load is lighter.

[noahbickart]

This is one the best ts threads in a while. Thanks everyone, especially David.

For “what we do,” that is, PA concert taping, 20db is likely about right much of the time.

To that end, we tapers should perform Dsatz’s test on the usual taper preamp suspects, at 20db.

[Gutbucket]

"Thus there exists no one gain setting for a typical analog preamp that both accommodates the maximum SPL of the microphone without clipping, and that simultaneously offers the lowest-noise performance for the quietest sounds that the microphone can pick up."

This brings into question how all the recent 32bit-floating-point recorders operate, each of which have integrated analog preamps of some sort or another which must accommodate a wide range of standard microphones of various sensitivities and output levels. In light of the above statement, accommodating such a range of diverse inputs would seem to be a considerably more complex challenge than the analog>ADC hand-off inside a digital microphone, where the output parameters of the specific microphone capsule and the ADC in the amplifier body can be closely matched.

"Secondly, (ca. 29:50) combination analog mike preamps and A/D converters exist that use multiple analog gain stages and gain ranging--and he says that such equipment can produce results that are very nearly equivalent to digital microphones."

I suppose this hints at is how its done.  Up until this point I've conceptually thought of those recorders as auto-ranging across multiple ADCs after a traditional preamp front-end stage, but had not considered that the auto-ranging is likely occurring ahead of multiple preamp stages, each specifically matched to its own ADC to optimize noise performance.

[wforwumbo]

Just piping that given I've been deep in the guts of a number of code bases that include metering, without going into detail and to respect individual companies' IP I wholly reinforce everything that DSatz says.

There's plenty of unknowns in implementation and too many standards (some for audio, some for film, some for television, some for game design) that all conflict with each other. I'm of the opinion that we are most likely to make a clean tape by setting levels conservatively at the preamp, and then raising levels digitally in post.

To reinforce a thought that keeps popping up in this thread, I'll comment that I frequently run my Omnis - which happen to be DSatz's former mk2 pair - with 20 dB of gain at the preamp, then I raise levels in post. It almost always makes a clean tape I'm proud of hearing without clipping, and I don't miss the added dynamic range at the lower end of the scale from setting levels so conservatively.

One additional monkey wrench I'll throw into this, is that I expect that all other things being equal (mic body, preamp, and digital gain at the bit bucket), if I were to run my mk4 cardioids vs my mk2 Omnis pointed at the same source I would expect that from the difference in frequency response the preamp would see less energy at its input for the cardioid. I suspect the best way to compensate for that is simply increasing gain, but I think it's foolish to run things through the mathematics for "proper level setting and gain compensation" instead of relying on our guts, intuition, and experience as recordists. And that's coming from someone who's known to be overly dependent on the scientific method, to a fault.

An additional note, to reinforce something DSatz has said above: "So I stick with the accustomed practice of maximizing the signal levels at each step along the way while taking care to avoid overload. Nothing fancy, just basic principles." This mirrors what analog electronics theory tells us, and is commonly referred to as "gain staging". The platonic ideal of electrical energy theory is to maximize power transfer from one section to the next, with as little distortion as possible. The theoretical ideal of any amplifier is to have infinite gain, infinite input impedance, and zero output impedance. In practice that's not exactly realizable (and actually not necessarily useful); so every analog design engineer worth their weight in gold that I know of, tunes amplifiers to match impedance to sections before/after it to maximize power transfer between sections, and tunes gain such that distortion can at least be measured and expected. In practice for the rest of us, what DSatz says above is what I think the best path forward is.

[Gutbucket]

It almost always makes a clean tape I'm proud of hearing without clipping, and I don't miss the added dynamic range at the lower end of the scale from setting levels so conservatively.

Typo?  More conservative levels will tend to reduce total available dynamic range.

This certainly bears repeating- "So I stick with the accustomed practice of maximizing the signal levels at each step along the way while taking care to avoid overload. Nothing fancy, just basic principles."

[rocksuitcase]

An additional note, to reinforce something DSatz has said above: "So I stick with the accustomed practice of maximizing the signal levels at each step along the way while taking care to avoid overload. Nothing fancy, just basic principles." This mirrors what analog electronics theory tells us, and is commonly referred to as "gain staging". The platonic ideal of electrical energy theory is to maximize power transfer from one section to the next, with as little distortion as possible. The theoretical ideal of any amplifier is to have infinite gain, infinite input impedance, and zero output impedance. In practice that's not exactly realizable (and actually not necessarily useful); so every analog design engineer worth their weight in gold that I know of, tunes amplifiers to match impedance to sections before/after it to maximize power transfer between sections, and tunes gain such that distortion can at least be measured and expected. In practice for the rest of us, what DSatz says above is what I think the best path forward is.

basic principles    YES!
on the topic of gain staging, not electronics as discussed above, but gear. my very first recording rig was Beyer M201c Hypercardioids and a Sony TC-D5M (1982). My mentor took the specs pf the Sony D5, focusing on its' input impedance. Then looked at the output impedance of several microphones we were choosing from. (Sennheiser 421, Sennheiser 441, Shure SM58's, Beyer M88, Beyer M201, Beyer M160). He advised us to buy the Beyer M201's because their output impedance was almost an exact match of the Sony's inputs. He told us that combo of mics/deck should almost never distort until we reached the max input SPL of the Beyers. (may have been 126dB???). I notice listening back that we RARELY had input distortion on those M201 tapes.
These were the stock pre-amps of the TC-D5M. Three years later, in 1985, we had the OADEs customize the inputs on three of our D5's.
Most of us are aware, but in summary for those who are referencing TS for technical reasons: levels can peak for a number of reasons, sometime overloading pre-amps and creating distortion. "Paying attention" to all gain stages in a recording chain is important. Utilizing most typical audience taper style pre-amplifiers gain at "medium-20-25dB" is probably ideal, as mentioned, one can increase gain at the recorder or mixer after that point in the signal chain. Microphones and pre-amps have various input and output impedance which should be analyzed before matching specific microphones with specific pre-amps.

[capnhook]

Re: Is levels peaking at 0 to(sic) high?







No

[Gutbucket]

Kevin bringing it back to ground level. Right. Less Than Zero is an Elvis Costello song.

Higher than zero is too high.
^
Print that on a T-shirt!

[Gutbucket]

D Satz, can you please help us with this?

Correct me if I'm wrong (I'm getting in above my head here), but..

The platonic ideal of electrical energy theory is to maximize power transfer from one section to the next, with as little distortion as possible.

Except this application is not in need of efficient power transfer, but rather voltage transfer with as little distortion as possible.  As I understand it, matching impedances achieves clean, efficient power transfer.  But with most audio equipment where voltage transfer is of interest rather than power transfer, a output impedance lower than the input impedance is desirable, no?

The theoretical ideal of any amplifier is to have infinite gain, infinite input impedance, and zero output impedance.

^ This implies the voltage transfer model of low-output-impedance>high-input-impedance rather than power transfer model of matched impedance, no?

[snip..] My mentor took the specs pf the Sony D5, focusing on its' input impedance. Then looked at the output impedance of several microphones we were choosing from. (Sennheiser 421, Sennheiser 441, Shure SM58's, Beyer M88, Beyer M201, Beyer M160). He advised us to buy the Beyer M201's because their output impedance was almost an exact match of the Sony's inputs. He told us that combo of mics/deck should almost never distort until we reached the max input SPL of the Beyers. (may have been 126dB???). I notice listening back that we RARELY had input distortion on those M201 tapes.

Just looked up the spec sheets for each of those microphones and all of them state an impedance of 200Ω. Well, except for the SM58 which Shure states as: rated at 150Ω, 300Ω actual, so effectively identical to the others, when compared to..

The TC-D5 spec sheets I found don't state the microphone-input impedance, but list line-in impedance as being: either 47kΩ or 100kΩ (two different spec sheets), elsewhere I found mic-input impedance reported as being 7.6KΩ or 4.6KΩ for the pro model with input transformers.  In any case, a significantly higher input impedance than the microphone's output impedance.

[edit] ..which seems to fit the voltage transfer model of low-output-impedance>high-input-impedance.

[DSatz]

Gutbucket, could we maybe take the discussion about impedance matching and power transfer, etc., into a different thread? It's more a historical issue (especially the huge influence of telephone technology on early professional audio in the U.S.) than a practical one nowadays, and it's almost entirely separate from level-setting in digital recording. Plus it's well settled; there isn't any real controversy or difference of opinion involved, that I'm aware of at least. Well, maybe some manufacturers of $200/foot speaker cable have their own unique concepts of physics ...

You've definitely got the basic ideas straight, though. All analog audio connection-making since some time in the 1940s or 50s has been based on voltage rather than power transfer, unless the signal is encoded as RF or the source impedance is very high (condenser microphone capsules!) or non-resistive in character (condenser microphone capsules, tape heads and phono cartridges). Except for the few remaining objectors to woofer damping, power amplifiers aren't impedance-matched with loudspeakers in domestic hi-fi systems the way they used to be in the vacuum tube era. And even balanced, analog audio connections in studios use bridging loads rather than matching ones, now that nearly 100 years have passed since Western Electric's system for electrical recording of phonograph records was introduced.

By the time of the TCD-5[M] the only real concern with those dynamic mikes would be that a few of them, including the Beyer M 201 if I remember correctly, and certainly the Sennheisers, were available in alternate models with different output impedances. German tape recorders still had DIN sockets with ca. 3.4 kOhm input impedance, and some people used "medium-impedance" microphones (ca. 600 Ohms actual output impedance) to drive those, at signal levels intermediate between what we now think of as mike level and aux or line level. But as long as you had microphones with 150-200 Ohm output impedance, IIRC the inputs of the TCD-5 were about 1.5 kOhm, so no problem there; the decision should have been based instead on the directional pattern and sonic characteristics of the microphones.

That said, the Beyer M 201 (I keep saying "Beyer" to distinguish it from an early Schoeps mike with the same model number) was a nice, non-hyped-sounding hypercardioid, sturdy and reliable. I think they still make a version of it, though they may have reduced its bass response when they introduced the "TG" series; I could never quite get straight information about that. -- The real sonic gem IMO among the microphones you listed was the M 160, though with its very low sensitivity (like only 1 mV/Pa) it wasn't always a practical choice. Given its weak signals and the unbalanced inputs of that cassette recorder, it would surely have had lots of RFI problems, and even more so in more recent times.

But if you want to carry this on any further, let's please move it to a different thread.

--best regards

[Gutbucket]

Thanks for that clarification, and no need for a more in depth discussion on my part.  Was questioning a few things the others mentioned  that didn't make sense to me, yet didn't feel I could speak authoritatively about them. Your understanding of these topics is considerably deeper than my own. 

I hope no one takes my any of my questioning posts above as confrontational, my intent is a clearer understanding for us all, including future readers of the thread. 

Since I have your ear, can you also clarify the bit below which got buried by other discussion that followed?  Big thanks for your participation here and your always excellent explanations which help make TS such a great resource.

~Lee

Following along. Mr Satz, can you clarify a couple lines in your post above for me?  You said-

"..especially if your recordings are ever played back in analog." [and later..] "All told, to me it seems very likely that there would be some s/n benefit if you raised your levels at least somewhat, especially if you consider the noise of your recorder's (or any media player's) analog output circuitry."

Its the statements after "especially" that caught my eye.  Does that assume analog transfer of the raw recording from the recorder? Or that a digitally transferred recording will not have levels adjusted in the process of producing the released recording?  If so, I see how what you describe can be the case.  However, most tapers are digitally transferring the raw recording to computer and adjusting levels in editing software as part of the process of producing the finished output they and others will listen to.  That process typically includes increasing signal level to remove excess recording headroom that no longer serves a useful purpose in the released version, and instead becomes a potential detriment for the reason you describe.  When that is the case, I fail to see how analog playback or analog output circuitry noise would effect a recording originally made with excess headroom differently than one made with optimal recording levels.  I understand how the first may potentially have a lower s/n due to a higher noise floor, but that would be baked into the recording and independent of playback.

Just making sure I'm understanding you correctly and not missing something there.

Thanks for the discussion - in particular your mention of the different noise spectra of condenser microphones in comparison to preamps and the noise characteristics of preamps at various gain settings.

[DSatz]

I said earlier that I have a hard time believing that a recording with very low peak levels can be relied upon to be as quiet as one made with more conventional levels. It's not impossible--but it depends on a whole slew of particulars, none of which I can ascertain in anyone else's recording equipment. I totally get it about setting conservative levels, but at some point there's a price to pay in signal-to-noise performance, and I want to know where that point actually is for my microphones and my preamp and/or recorder rather than guessing.

From what I've seen in numerous messages on this board, I think some people's intuitions about gain and noise aren't how they actually work. Some people seem to think that the noise added by a gain stage will be, in essence, directly proportional to the amount of gain being added, so that lower gain settings will translate more or less directly into less noise. I've seen people say, for example, that for reasons of noise they'd rather connect a (relatively high-output) microphone to a recorder's line input than its mike input even if they have to turn up the record level control all the way and their recorded peaks don't get anywhere close to 0. They count on 24-bit or 32-bit recording to make up the difference. I'm pretty sure I remember seeing this w/r/t the Sony M10 recorder, for example. Unfortunately as far as I was able to tell when I tested one (not in a very elegant way), the 24-bit setting was hardly quieter than the 16-bit setting--there was less than a 2 dB difference as I recall. Now, I wasn't completely confident of my measurement result in that case. But if it was right or nearly so, then the option that the person chose based on their concept of how it should work, was probably a bunch noisier than the alternative that they rejected.

It's hard to do valid comparison testing in live recording situations; that's more of a test-bench project. Some kind of tone generator and voltmeter, separate from your recorder, are essential. And preferably there should be one other thing on hand that's less common, because when you connect a tone generator to the input of a mike preamp, the preamp doesn't "see" the same source (driving) impedance as that of your microphones, which in turn affects the noise floor. So it really helps if you have the kind of test setup, for the particular brand and type of microphones that you use, that a service technician would use if they work seriously on that kind of microphone. For condenser microphones with removable capsules, manufacturers make shielded "test heads" (a/k/a measurement adapters) that can be attached in place of a capsule, and some manufacturers will sell you one if you ask nicely. A test head places the same capacitance across the microphone amplifier's input, but you can feed signals through it, or you can short its input to measure noise (which still leaves the capacitance in place). The microphone amplifier, meanwhile, is powered in the normal way, so everything in the circuit is just as it would be in a real recording setup.

I have a test head like that for the type of microphones that I usually use (no secret--Schoeps Colette series), and I used it for my preamp noise tests. I set the preamps to 30 - 35 dB gain and the results that I got were not IN ANY WAY predictable from the spec sheet values for those preamps, which were all compiled at higher gain settings. I know of no way to predict or infer the noise performance of a preamp at a given gain level (and microphone source impedance) from knowing its noise performance at its highest gain setting with resistors across its inputs. I've never measured preamp noise at lower gain settings, but other people's measurements seem to indicate that for the best noise performance from typical preamps, you generally want a setting more in the direction of the highest gain that you can set without overloading anything--not the lowest.

So the remaining question is how that's affecting person X's recordings with recorder Y and mike preamp Z using microphones A and B, which I can't answer from here, not having their set of gear and the test equipment in front of me. But the assumption of getting similar (let alone better) noise performance from a preamp at low gain than with the same preamp set to a higher (but still safe) setting, isn't a safe assumption. It's definitely wrong in at least some cases, perhaps in many or even most, but I'd want to see a lot more measurement results before generalizing that far.

Thus my uncertainty, but thus also my sense that somewhat higher gain earlier in the "recording chain" could bring an improvement in signal-to-noise ratio without necessarily taking away safety.

[Gutbucket]

Thanks. A couple follow up questions:

Is is possible to make any generalizations about the different noise spectra of condenser microphones vs. preamps vs. the acoustic noise floor of a quiet recording environment, that might help folks unable to make noise measurements discern which of those is the primary contributor to the noise-floor in a recording they make?


In an effort to maximize s/n in the absence of actual noise measurements, I fall back on the practice of trying to achieve the lion's share of needed gain at the forward-most end of the signal chain, followed by setting recording levels to leave sufficient yet not excessive headroom to accommodate expected and somewhat predictable SPL increases and unpredictable peaks.

The first part of that is mostly about setting things up initially. And I've assumed more sensitive microphones helpful in that the microphone itself represents the earliest gain stage in the signal chain, thus requiring less gain from the preamp stage that followings (outboard or in the recorder). All good as long as the microphone's output is not so high that it overdrives the following stage.

The second part about setting gain levels is less predictable and involves more variables, including the type of music, specific program material, distance to the source, and in the case of many TS members, additional constraints such as not being able to monitor or adjust levels during the performance.  Some situations are going to require significantly more headroom than others, and the key becomes determining the Goldilocks amount - not too much, not too little.

Do you find this to be a reasonable approach?

[DSatz]

Condenser microphones have three or four main noise sources: the input noise of the first active device in the amplifier circuit, the capacitive source impedance of the capsule, and random air motion around the diaphragm. In "traditional" condenser microphones there's also some noise from the very high-value resistor through which the capsule is charged (polarized). Mike preamps have mainly the input noise of their first active device, but also the noise from the microphone including that of its source impedance.

These noise spectra tend to increase gradually toward higher frequencies, roughly like pink noise, except for the capsule. Its source impedance is highest at lowest frequencies, basically "1/f", and its level is quite significant--billions of Ohms at the bottom of the range. But its audible effect depends on subsequent amplification, since our hearing is so insensitive at low frequencies and low SPLs.

So if you look at a swept-frequency graph of noise output of any condenser microphone in a silent room or enclosure, going from the lowest to highest frequencies you'll see first the capsule noise with its downward slope, crossing over (normally somewhere on the left side of the graph) to the rising, more-like-pink noise of the other sources. The greater the capacitance of the capsule, the lower both the frequency of this transition and the amplitude of the noise will be, "all other things being equal". IIRC the transition is generally below 1 kHz for most studio microphones, sometimes well below. Maybe not the ones with ultra-small capsules, though.

As you can well imagine, with such an uneven distribution of noise across the frequency spectrum, any frequency weighting that you use will have a huge effect on the result; likewise the averaging (integration) over time intervals, which tends to erase the impulse noise that is by far the most disturbing to listeners. This complexity tends to cloud discussion, and to produce frustration among those who want simple answers to reasonable questions such as "how much noise does this microphone produce?". Unfortunately there's no easy answer.

--In your description of how you approach "gain staging" overall, the only assumption I question is that more sensitive (higher-output) microphones -> generally more desirable. That's only a "sometimes" thing. As we've been discussing, it's not necessarily productive to try to relieve a good mike preamp of its duty, and it can be counterproductive.

--best regards

[capnhook]

This complexity tends to cloud discussion, and to produce frustration among those who want simple answers. Unfortunately, there are always people ready to offer simple answers to complex questions. I wish that people wouldn't buy those answers so readily, here or in the world outside of sound recording (which I'm told exists).

https://taperssection.com/index.php?topic=202622.msg2400252#msg2400252

[Gutbucket]

Thanks, that helps.  I certainly wish to avoid the oversimplification of any conclusions, as well as the beating of a dead horse discussion into the ground of which I may already be guilty..

In the gain staging approach I mention above I'm assuming the use of preamps built into the recorder, and/or one of the small external preamps typically used for small unbalanced microphones.  The quality of such preamps has improved over what tended to be found in "prosumer" recording gear of years past, yet is nothing world-class. 

Since the external preamps you regularly use and have tested "in-the-chain" are presumably of better quality than the ones I'm using, yet were all found to have noise characteristics that must be measured to be accurately determined, in the interest of maximizing s/n without such measurements available to me, I default to choosing the higher sensitivity option between two otherwise identical micrphones as long as the higher output isn't going to overdrive whatever stage that follows.
^
My take away from the discussion here is that this presumption is insufficient in itself, yet without figuring out some way to do the actual testing, is at least reasonable when combined with setting the gain of the adjustable preamp stage which follows somewhere around the middle of its range and not overly low or high.  This is somewhat analogous to guessing at the appropriate RPM range at which to shift a manual transmission, without knowing the actual torque curve of the engine, based on the general shape of most torque curves.


The new 32-bit world may change this if what is going on under the hood is auto switching between multiple paralleled preamp stages with  each having been tuned for maximum s/n within its level window.