Binaural bass crackling: How to fix/avoid?

[Greg Miller]

Micky, I've posted the image files elsewhere, hopefully I'll be able to link to them, so here goes...


Distortion, of course, can take many forms. A very common form of distortion is clipping, caused when the signal amplitude is greater at some point than the signal chain can handle; the signal chain "runs out of headroom" for that signal.

In your demo file, we could call the "clicks" a type of distortion (although we could also call them a type of noise). We ultimately will try to inspect the waveform of your file visually, in order to determine whether the "clicks" are typical clipping or not. That might give us a clue as to how to avoid this problem in the future.

First let's have a quick demo of clipping.

demo1-1.gif
Demo1-1.gif shows a few syllables of speech, without distortion (this is from a file I had handy). The top green waveform is the left channel; the bottom waveform is the right channel. I've marked a few reference levels (on the right channel only): 100% (0dB), and 50% (-6dB). Note, too, that I've highlighted a small portion in the second syllable.

demo1-2.gif
Demo1-2.gif is the same file, but now I've zoomed in on the portion that was highlighted. Note that the peaks (the loudest levels in this part of the file) have a rather smooth, rounded shape.

demo1-3.gif
Demo1-3.gif shows the same timeframe as the first image. Now, however, I have raised the gain by +6dB, so the signal amplitude is twice as loud. Compare the second syllable here (with the highlighted area) with the second syllable on the original image (Demo1-1.gif). Here, all the peaks stop at a level of 100%, because, by definition, the signal chain can not handle more than 100%.

demo1-4.gif
Demo1-4.gif zooms in on the area highlighted in the previous image. This clearly shows an example of clipping. When the waveform reaches 100% (the thin horizontal gray line), it can't go any higher. So, although the peaks originally were rounded (Demo1-2.gif), they now become flat-topped. This is clipping.

demo1-5.gif
Demo1-5.gif shows that you can't easily fix a clipped signal. Here I have taken the clipped signal from Demo1-4.gif, and lowered the gain by -6dB. Thus, most of the waveform is back to the original level. But compare the peaks here to the nice rounded peaks in Demo1-2.gif. Now they are flat-topped, even though their peak level is -6dB. That's because they were clipped previously (Demo1-4.gif) at 100%, and when we reduce the gain by -6dB (50%), they stay flat-topped but at a lower level. They are still clipped, and if you have a significant number of clipped peaks like these, the clipping will be quite audible.

In fact there is some software that will try to restore clipped peaks, and it will more or less work in some cases. It would probably work for the small amount of clipping in this demo. But that's beside the point for the sake of this discussion. The point of this demonstration is simply to show what a clipped waveform looks like.

In the next installment we'll take a look at your waveform and compare your audible clicks to this illustration of clipping.

---

PS: Micky, I'm sorry, I cannot get those images to display here. I did put the full URL in the body of my message, and the BBS software converted it to a "url=" format, so you have clickable links on the page. I guess you'll have to click the link, then <alt><tab> back and forth between my text and the relevant image. ... Anyway, I hope this description and visual depiction of clipping provides an introduction for you. As I said, the next installment will deal with the file you sent that has the strange clicks in it. Meanwhile, I will continue trying to figure out how to display images within the body of the message.

[Micky Hulse]

Greg....

Awsome!!!!!

Thanks soooo much for all of your help! Very very very informative! I am learning a ton. This is great because I do plan on recording a bunch more with my binaural mics... This will definitely help me out for when it comes to editing (same with audio from my video clips!)

Thanks man! I really owe you one!!!! :)

Originally Posted by greg miller

...<snip>...
demo1-5.gif
demo1-5.gif shows that you can't easily fix a clipped signal. Here i have taken the clipped signal from demo1-4.gif, and lowered the gain by -6db. Thus, most of the waveform is back to the original level. But compare the peaks here to the nice rounded peaks in demo1-2.gif. Now they are flat-topped, even though their peak level is -6db. That's because they were clipped previously (demo1-4.gif) at 100%, and when we reduce the gain by -6db (50%), they stay flat-topped but at a lower level. They are still clipped, and if you have a significant number of clipped peaks like these, the clipping will be quite audible.

That's all so interesting!

What program are you using?

Just to clarify, in demo1-5, did you save the clip from demo1-4, and then re-open it? I assume this is the case, I just wanted to make sure this is the case.

Originally Posted by greg miller

In fact there is some software that will try to restore clipped peaks, and it will more or less work in some cases. It would probably work for the small amount of clipping in this demo. But that's beside the point for the sake of this discussion. The point of this demonstration is simply to show what a clipped waveform looks like.

Interesting! Of course, my goal would be to capture the best sounding audio from the source, but that is still good to know that there are softwares that can fix stuff like that. Do you happen to know of a Mac software that does this?

Originally Posted by greg miller

In the next installment we'll take a look at your waveform and compare your audible clicks to this illustration of clipping.

Man, I am totally stoked! This is soo cool. Thanks so much for taking the time to school a noob like me! :)

Originally Posted by greg miller

ps: Micky, i'm sorry, i cannot get those images to display here. I did put the full url in the body of my message, and the bbs software converted it to a "url=" format, so you have clickable links on the page. I guess you'll have to click the link, then <alt><tab> back and forth between my text and the relevant image. ... Anyway, i hope this description and visual depiction of clipping provides an introduction for you. As i said, the next installment will deal with the file you sent that has the strange clicks in it. Meanwhile, i will continue trying to figure out how to display images within the body of the message.

Argh, I know! I wish this forum would at least let us embed images... I think the "url=" format is the best we are gonna get. :(

Actually, I do have a question about what Paul said:

Just a thought - but did you use the battery module? The reason for asking is that if the bias voltage is supplied by the recorder, then the dynamic range of the mic is quite low - 105dB SPL (120 db when used with one of our battery modules). With so much bass, it's quite possible that the distortion are the mics itself - the recorder recording the distortion faithfully, which might also explain why the file, without being tinkered with, is at a modest level, but showing these clicks. The waveform, as mentioned hasn't flat topped, but is just spiky and rough - there's no audio peaks that cause it like snares or other percussive stuff - so could be the heavy bass components below 50Hz - which on a spectrum display seem very prominent.

I hope this does not sound like a silly question, but what does he mean by "bias voltage"? Is that phantom power?

How often in day-to-day life do we encounter bass below 50Hz? Could wind rumble reach such frequencies?

Also, I will have to compare my waveform to your example images... Paul mentions "spiky" and I noticed your non-clipping examples show a smother/rounded waveform at the peaks...

Hrmm, sorry, I am rambling! When I get home from work later today I will have to piece all of this together. This is all very interesting!

Thanks again Greg!!!!!!! Much appreciated!!!!

Have a great day!

Cheers,
Micky :)

[Greg Miller]

Hi Micky,

To answer one of your first questions: Actually I never saved that demo file at all. The original file was something that another member (Syed Junaid) had posted on a different website's forum. (He posted a lot of the same questions on that forum and on this one.) It was speech with a lot of background noise, but the speech was *not* distorted. I happened to have that file handy so that's what I used.

So I opened it, made a few screen captures, applied +6dB of gain, captured again, applied -6dB of gain (in other words, 6dB of gain reduction), and made the final capture. Then I just closed the file without saving it, because I had no reason to save it.

--

I work on the PC platform exclusively. For this demo, I used Cool Edit Pro (v. 2.something). That's no longer on the market. Adobe bought the company (Syntrillium) who wrote Cool Edit, and re-badged it as Adobe Audition. Audition has evolved through a few versions, and has some additional features not found in Cool Edit. You might occasionally find an old version of Cool Edit for sale on eBay, but AFAIK it's PC only. I don't know whether Audition is also available for Mac, or not... it should be easy for you to find out. I don't know anything about any Mac software.

--

"Plug in power" is sometimes called "bias voltage" although that's really inaccurate... bias voltage is actually something entirely different. This system is used with consumer equipment that has unbalanced mic connections. Unbalanced means there are two terminals for a given mono mic, and two conductors in a given mono mic line: hot, and return/ground/shield.

In order for electrical current to flow in a simple circuit (one channel of audio, a flashlight, a table lamp, etc.), you must have two terminals. A loudspeaker is a good example of this, it needs two terminals, and you normally use two-conductor zip cord (lamp cord) to get the signal from your power amp to the speaker.

In theory, you could use the same type of zip cord with a dynamic microphone; but there would be problems. The microphone preamplifier in your recorder (or mixer, etc.) has very high gain, in order to boost the very weak signal coming from the mic. If you used zip cord on your mic input, that wire would act as an antenna and would pick up all sorts of electrical noise... strong hum from your house's power wiring, clicks and pops from appliances turning on and off, maybe strong nearby radio stations, too. Therefore, for a mic input you need to use "shielded" cable... look it up on Wikipedia. The common ground circuit of the preamp is connected to the shield at the mic jack, and the common ground end of the mic is connected to the shield at the other end. So in this case, the shield of the wire also acts as a ground connection, and is also one of the two conductors carrying the mic signal to the preamp. This is called "unbalanced" because one of the two conductors (the "hot" one) is **not** grounded, but the other one (the "return") one **is** grounded. They are not the same, hence the circuit is UNbalanced.

Condensor mics have active electronics in them... at the very least a single FET that acts as a type of preamp (actually an impedance converter). This electronics needs DC voltage to operate. Of course it could come from a battery inside the mic housing, and some mics work that way. But it's also possible to send the DC voltage to the mic, from the preamp, over the mic wiring. Consumer equipment uses the "plug in power" scheme. This is done by connecting a low DC voltage source (usually between 1.5 and 9 volts) from the preamp, through a resistor (usually around 1,000 ohms or a bit more) in the preamp, to the "hot" terminal of the mic wiring. Then that DC voltage uses the "hot" mic wire to get to the mic, where it powers the electronics. Eventually the audio voltage flows back to the preamp using the same "hot" wire. The ground/shield is the return conductor the the audio, and also the return conductor for the DC current.

Professional mics use a different scheme, which is "balanced" wiring. A dynamic element (which generates the mic signal voltage) has two terminals on it. In a professional dynamic mic, each terminal goes to an insulated conductor in the mic cable, and then on to two input terminals on the preamp (pins 2 and 3 of the XLR connector). In addition, there is a shield, which connects the ground of the preamp to the metal body of the mic. But in this case, there is NO connection between the mic element and the shield... there are three entirely separate conductors (two inside, plus the shield). And NO audio current flows through the shield. The braid of the shield protects the two audio conductors (which are inside) from electrical and RF noise in your house, and that's all it does. The two audio conductors are both similar, in that neither one of them is connected to shield or ground at any point. They are the same, hence the audio signal is "balanced."

With a professional condensor mic, a higher DC voltage (sometimes as high as +48 volts) is fed from the mixer, equally on both of the audio conductors (pins 2 and 3), to the mic. The negative side of the DC power supply is connected to the mixer's ground (same as the negative terminal on your car battery is connected to the chassis), and the DC current from the mic's internal electronics returns to the mixer through the shield of the mic cable (pin 1). This is "phantom" powering.

As you see now, "plug in power" and "phantom" are entirely different. The former applies only to UNbalanced mics, the latter applies only to balanced mics. The former uses a fairly low voltage (+9 volts or less) while the latter uses a higher voltage (usually at least +12 volts, and often as high as +48). "Plug in power" and "phantom" are entirely non-compatible. Mis-connecting something could easily damage a mic. So you need to always keep in mind which one you're using. A consumer-style recorder, with a 3.5mm mini jack for the mic connector, would be "plug in power." A professional recorder or mixer would be phantom. DO NOT MIX!!!

--

We encounter very low frequencies all the time in day-to-day life. If you're in a room and someone opens or closes the door, that will produce a positive or negative air pressure peak, and, depending on the room dimensions, there may be a few cycles of decaying resonance. It's below the range of audible frequencies, and you don't need or want to record it. Wave your hand past your ear, and you'll generate a very low frequency pressure wave, but again it's sub-audible and you don't need or want to record it.

I used to work at a radio station that had some awesome RCA ribbon mics. When it was time for the station to sign off at midnight (this was *many* years ago), I would hold a broom out at arm's length and swing it past the mic (with the mic turned on). The low-frequency voltage pulse from the mic made it through the mixing board, through the audio processing, and into the transmitter, where it caused such a huge overload that the transmitter's safety circuits would shut down the transmitter! Sub-audible energy at work!

Music contains a fair amount of energy below 50 Hz, and if you need to roll off it's preferable to do so below 30 Hz. Put on your some headphones (a Sennheiser 280 or better) and listen to a well-recorded classical CD. Listen to the tympani... you can feel them. Listen to the "room tone" between tracks, when nobody is playing their instruments; you will hear and almost feel the low frequency resonance of the room. On the other hand, for dialog recording, you can easily roll off at 50 Hz, or perhaps even higher if you have a lot of background noise to deal with. But by the time you get up to 100 Hz you're starting to affect the voice quality... it will still be completely intelligible, but it will start to become unnatural. Of course with higher-pitched female voices you can move the rolloff up a little higher in frequency.

--

The only waveform that is completely perfectly rounded is a pure sine wave. Such a thing rarely exists in nature. How a waveform looks depends on how far you zoom in. ***SPOILER ALERT*** If you zoom in enough on your own track, it will start to look more and more rounded. The key is that the peaks are not flat-topped. Zoomed out they look "spiky" and zoomed in they look rounded... they are not clipped. If they were clipped there would be a huge number of flat tops, and your track doesn't have that.

Is Micky's track clipped?

What are those clicks?

Tune in again, same time, same station, for another episode of "Cosmic Clicks"!

[Micky Hulse]

Greg, this is like a master class in audio. Amazing stuff!!!!

Do you have an Amazon wishlist? I would love to get you a book or something. :)

Originally Posted by Greg Miller

...<snip>... So I opened it, made a few screen captures, applied +6dB of gain, captured again, applied -6dB of gain (in other words, 6dB of gain reduction), and made the final capture. Then I just closed the file without saving it, because I had no reason to save it.

Ahhh, great! Thanks for the clarification! I was just not sure if you saved before/after applying/removing the +/-6DdB... Thank you for the additional info. :)

Originally Posted by Greg Miller

I work on the PC platform exclusively. For this demo, I used Cool Edit Pro (v. 2.something). That's no longer on the market. Adobe bought the company (Syntrillium) who wrote Cool Edit, and re-badged it as Adobe Audition. Audition has evolved through a few versions, and has some additional features not found in Cool Edit. You might occasionally find an old version of Cool Edit for sale on eBay, but AFAIK it's PC only. I don't know whether Audition is also available for Mac, or not... it should be easy for you to find out. I don't know anything about any Mac software.

Looks like I could run Audition via an emulated windows environment... I will have to find a demo and give Audition a whirl on one of the PCs at my work. :)

Originally Posted by Greg Miller

"Plug in power" is sometimes called "bias voltage" although that's really inaccurate... bias voltage is actually something entirely different. This system is used with consumer equipment that has unbalanced mic connections. Unbalanced means there are two terminals for a given mono mic, and two conductors in a given mono mic line: hot, and return/ground/shield. ...<snip>... As you see now, "plug in power" and "phantom" are entirely different. The former applies only to UNbalanced mics, the latter applies only to balanced mics. The former uses a fairly low voltage (+9 volts or less) while the latter uses a higher voltage (usually at least +12 volts, and often as high as +48). "Plug in power" and "phantom" are entirely non-compatible. Mis-connecting something could easily damage a mic. So you need to always keep in mind which one you're using. A consumer-style recorder, with a 3.5mm mini jack for the mic connector, would be "plug in power." A professional recorder or mixer would be phantom. DO NOT MIX!!!

Wow! Like I said, master class!!!!

Tons of excellent information Greg! THANK YOU!!!!

Sorry if I mixed up my lingo. When shooting video, I use an XHA1 (Canon) and, IIRC, I think the manual calls the XLR +48 switch "phantom" power. I just assumed the "plug in power" on my R-09hr was the same thing. Thank you for taking the time to teach me the difference. :)

Originally Posted by Greg Miller

...<snip>... Is Micky's track clipped?

What are those clicks?

Tune in again, same time, same station, for another episode of "Cosmic Clicks"!

Lol! I am looking forward to the next installation!!!

Let me know if there is a book I can get you off of Amazon or something... If you setup an Amazon wishlist then I could have it shipped to you quick and easy. ;)

Thanks a billion Greg!!!!!

Have an excellent weekend. :)

Cheers,
Micky

[Greg Miller]

Micky,

Thanks for the kind words. I'm glad some of this info is useful to you.

Thank you also for the generous offer of a book. Nothing comes to mind, so don't worry about it. Besides, I'm not doing this with the thought of receiving any sort of "earthly goods." It's reward enough to know that what I'm doing is helpful to someone... in this case, to you. Besides, if I ever get back to Portland, you owe me a beer! ;)

Seriously, I believe in some sort of "pass it along" philosophy... hopefully some day you will be able to spend a few minutes and pass along some knowledge to someone else who needs some help.

Since you mentioned books, it occurs to me that you, yourself, might benefit from a few good books. I can recommend Jay Rose as a very knowledgeable audio guy and author. You can get some information about his books on his website: How-to books about Sound for Digital Filmmakers and of course if you back up to his homepage you will find more useful information. I have a copy of his Postproduction book and I think the information is pretty good. (Jay posts frequently on another Digital Video forum, but I haven't seen him on this one.)

Now as to your original question... As you might imagine, it takes a bit of time and concentration to put together a comprehensive post about this stuff. All well and good, I'm not complaining, it's good mental exercise for me. However, the last few days I've been fighting a pretty nasty toothache, and the Percoset has my head spinning... literally. So I don't think I will be composing any lengthy technical comments in the next few days, until the tooth situation is resolved. Sorry to ask you to wait any longer, but please be patient. I've not abandoned the question, I just think I'll do a better job with it when my head is clear.

For now... Happy Trails!

[Robert Wiejak]

Gentleman,
I will admit that I have not read the whole post, but I listen to your sample and found your ‘clicks’.
To me it looks like hardware, maybe power supply problem.
Take a look at the first picture and see where the click is in that waveform.



Then look at the second one and see if you agree that the waveform should look like the blue line in the second picture.



The whole ‘signal rise’ (the correction) in the first picture (the top waveform) is only over 7 samples @ 96Ksps, hence the click.
The difference over these 7 samples is 68dB.

I don’t have a solution for you, just thought I’ll share my findings. Interesting problem.

[Greg Miller]

@Rob

Yes, that's the pattern that we're seeing. It is not always 7 samples, sometimes it seems to be as short as 3 samples.

Originally Posted by Rob Wiejak

The difference over these 7 samples is 68dB

I think you're way off. For example, in one "click" I analyzed, the "pre-click" sample value was 3959127, the "post-click" sample value was 3337707. Dividing those numbers gives a result of 0.843, which corresponds to a change of -1.48dB. If you expand whatever waveform display you're looking at, the vertical (amplitude) scale of the display will confirm that.

Cause still to be determined...

[Robert Wiejak]

Originally Posted by Greg Miller

I think you're way off. For example, in one "click" I analyzed, the "pre-click" sample value was 3959127, the "post-click" sample value was 3337707. Dividing those numbers gives a result of 0.843, which corresponds to a change of -1.48dB. If you expand whatever waveform display you're looking at, the vertical (amplitude) scale of the display will confirm that.

Funny thing about dB's, they are very misunderstood creatures.

dB is a ratios of two (one known, the other measured) quantities expressed in logarithmic units.
If you ware referring to full scale (the known value), compared to distorted samples (pre-click & post-click = measured value) then you are absolutely right, the difference is less than 1.5dBFS.
But I didn’t say full scale, I said ‘…the difference over these 7 samples…’ implying same as you pre-click & post-click:

-14919 - -17513 = 2594

Then to get the voltage ratio of the difference: 20 * log(2594) = 68.28dB

I hope this explains how I arrived at that number.
Remember, dB’s are not absolute numbers, they are always in reference to something.
Your numbers are in reference to full scale, my numbers are in reference to each other (pre-click vs. post-click).
Both results are correct.

[Greg Miller]

Sorry, no, both results can't be correct, because 68 does not equal 1.5.

Originally Posted by Robert Wiejak

Funny thing about dB's, they are very misunderstood creatures

Funny thing is, I've been using dBs for 40 years, both in terms of circuit voltages, sample values, and in terms of antenna power. And I do understand the proper way to use dB.

When comparing two signals, you either divide the arithmetic sample values, or you subtract the dB values. The two operations, dividing the arithmetic values, or subtracting the dB values, are mathematically equivalent.

You're getting confused because you subtracted the two sample values, rather than dividing them to get their ratio.

As you correctly state,

Originally Posted by Robert Wiejak

dB is a ratios of two --<snip>-- quantities expressed in logarithmic units

dB always represents a ratio, which is then converted to a logarithmic scale.

Since your goal is to express how the second sample is related to the first sample, and express this relationship in dB, you need to start out with the ratio of the sample values. To get the ratio, you divide the two values. (Your mistake was that you did not find the ratio between the two values, because you did not divide them... you erroneously subtracted them. Subtracting does not give you a ratio.)

Then, as you state, you convert that ratio to a log, and multiply that log times 20.

The arithmetic sample values are absolute values. 3,959,127 and 3,337,707 are the actual value of those samples (converted from the binary data stored in the file). Full scale doesn't matter. Full scale could be any number bigger than 3,959,127. (Of course the actual value of "full scale" depends on the bit depth of the samples.) Regardless of what full scale is, those sample values are what they are.

In this case, to find the ratio of the two samples, you divide the second value by the first value, and you find that the second value is 84.3% of the first number. The log of 0.843 is -0.074. Multiply that by 20, and the result is -1.48dB. Therefore, the second sample is -1.48 lower than the first sample.

--

Don't take my word for it. (Apparently you do not.) You probably don't have the same books that I have in my reference library, so let's go to a readily available source: Wikipedia. Look up "decibel." Scroll down to the section titled "Field Intensities." (Note that the dB originally was used to measure power ratios, where the formula is 10*log(base10) of P1/P0.) Voltage, or in this case digital samples, is not a measure of power, it's a measure of intensity. So in this case the formula is 20*log(base10) of A1/A0.

Again, you did not divide A1/A0, you subtracted A1-A0. That does not conform to the correct formula, as shown above (from Wikipedia).

--

Let's look at it graphically. We'll use the file you posted. The "pre-click" sample you have circled is at -5.45dB on your software's scale. The "post-click" sample you have circled is at roughly -6.8dB. The second sample is about 1.35dB softer than the first sample. Incidentally, that's fairly close to the two samples that I analyzed... quite close, considering we analyzed two different samples at two different points in the file. In fact, the distortion mechanism here is quite consistent from one click to the next.

--

Another way to think about this: If our two "click" samples were 68dB different, that wouldn't be just a little glitch on the graph, it would be a huge jump. If the first sample was -5.45dB, the second sample would be -73.45dB. That wouldn't be just a slightly audible click, that would be a huge speaker excursion!

--

Look at it mathematically, look at it graphically to confirm it. Either way, the two samples differ by roughly 1.5dB, and that is not the same as 68dB. -68 is an entirely spurious number.

[Micky Hulse]

Originally Posted by Greg Miller

Thank you also for the generous offer of a book. Nothing comes to mind, so don't worry about it. Besides, I'm not doing this with the thought of receiving any sort of "earthly goods." It's reward enough to know that what I'm doing is helpful to someone... in this case, to you. Besides, if I ever get back to Portland, you owe me a beer! ;)

Hehe! Well, let me know if you change your mind. I totally don't mind getting you a treat off of Amazon... It's the least I could do to say thanks for all of your pro help. :)

Beer sounds good though!

Originally Posted by Greg Miller

Seriously, I believe in some sort of "pass it along" philosophy... hopefully some day you will be able to spend a few minutes and pass along some knowledge to someone else who needs some help.

That's a great philosophy! I believe in the same thing (I try to be as helpful as I can in the areas that have more experience in). It's true what they say: What goes around comes around. :)

Originally Posted by Greg Miller

Since you mentioned books, it occurs to me that you, yourself, might benefit from a few good books. I can recommend Jay Rose as a very knowledgeable audio guy and author. You can get some information about his books on his website: How-to books about Sound for Digital Filmmakers and of course if you back up to his homepage you will find more useful information. I have a copy of his Postproduction book and I think the information is pretty good. (Jay posts frequently on another Digital Video forum, but I haven't seen him on this one.)

Awsome! Ordering

both

 

both

of Jay's

books

 

books

now. Thanks for linkage. :)

Originally Posted by Greg Miller

Now as to your original question... As you might imagine, it takes a bit of time and concentration to put together a comprehensive post about this stuff. All well and good, I'm not complaining, it's good mental exercise for me. However, the last few days I've been fighting a pretty nasty toothache, and the Percoset has my head spinning... literally. So I don't think I will be composing any lengthy technical comments in the next few days, until the tooth situation is resolved. Sorry to ask you to wait any longer, but please be patient. I've not abandoned the question, I just think I'll do a better job with it when my head is clear.

No worries! I completely understand. No rush at all. You could stop now and I would still feel like repaying you with an Amazon treat! :D

Originally Posted by Greg Miller

For now... Happy Trails!

Ditto! Thanks Greg!!!!

Have an awsome day!

Cheers,
Micky

[Micky Hulse]

Hi Robert! Many thanks for the help! I really appreciate it. :)

First, how the heck did you get embeded images inline with your your post!!! I see you are using "attach" and then the ID of the attachment. Pretty cool! I will have to play around with this the next time I want to post images inline with my comments. :)

Originally Posted by Robert Wiejak

Gentleman,
I will admit that I have not read the whole post, but I listen to your sample and found your ‘clicks’.
To me it looks like hardware, maybe power supply problem.
Take a look at the first picture and see where the click is in that waveform.

Ugh, I would hate for it to be hardware related. :(

I can't really afford to re-buy a recorder and/or binaural mics.

Do you think it could be the battery pack?

In the last tests I did at home I did not experience any cracks/pops, but then again the bass was not as deep, although I did have my head right up next to the speaker (and it was turned up loud... My wife kinda got pissed at me that day). :D

Originally Posted by Robert Wiejak

I don’t have a solution for you, just thought I’ll share my findings. Interesting problem.

Thanks so much for helping out!

I think the next time I am down in S.F. (should be a couple months) I will go back to the planetarium to re-record; I will be sure to note all of my settings this time.

Thanks Robert!!! Much appreciated! :)

Have a great day!

Cheers,
Micky

[Steve House]

Originally Posted by Micky Hulse

..

I think the next time I am down in S.F. (should be a couple months) I will go back to the planetarium to re-record; I will be sure to note all of my settings this time.
..Micky

Why re-record? The clicks are not all that objectionable if you're simply listening for your own enjoyment and since I'm sure the program is copyright, personal listening is all you'll ever be able to do with it legally.

[Micky Hulse]

Originally Posted by Steve House

Why re-record? The clicks are not all that objectionable if you're simply listening for your own enjoyment and since I'm sure the program is copyright, personal listening is all you'll ever be able to do with it legally.

Good points! :)

I personally don't want to see it again, but it's the only situation that I can think of with bass that deep.

Also, I was thinking that it would be nice to compare the same audio, but with the second one I would have noted all of my recorder settings. My main concern is avoiding pops and clicks for future recordings that are NOT in a planetarium! :D

Last time I was in California, I spent every day photographing, gps logging and audio recording (almost) every day of my vacation. This may sound cheezy, but I was doing it in the name of "art" and "journalism" (i.e. documenting my vacation). I ended up with a ton of cool audio (the Dim Sum and F-line Streetcar recordings were some of the most interesting audio captures).

Unfortunately, the only time I noticed the clicks was during the planetarium event, otherwise I would have chosen another audio clip to share on these forums. :)

Thanks for the reply! Much appreciated!

Cheers,
Micky

[Steve House]

Originally Posted by Micky Hulse

...Last time I was in California, I spent every day photographing, gps logging and audio recording (almost) every day of my vacation. This may sound cheezy, but I was doing it in the name of "art" and "journalism" (i.e. documenting my vacation). I ended up with a ton of cool audio (the Dim Sum and F-line Streetcar recordings were some of the most interesting audio captures).

...

I lived many years in San Francisco in the days when the green and yellow PCC cars were the backbone of the Muni trolly lines, worked right at the corner of Market and Van Ness were a number of the inbound cars reversed on the wye at Market and 11th. I lived for a while on Church Street with the streetcars running right past my front door. Their sounds were a constant accompaniment to my day, both at home and at work, and when I hear them now it makes me very nostaligic. The same with the sound of the foghorns (shut down back in the 80's) echoing in from the bay late on a damp and foggy night.

[Micky Hulse]

Originally Posted by Steve House

I lived many years in San Francisco in the days when the green and yellow PCC cars were the backbone of the Muni trolly lines, worked right at the corner of Market and Van Ness were a number of the inbound cars reversed on the wye at Market and 11th. I lived for a while on Church Street with the streetcars running right past my front door. Their sounds were a constant accompaniment to my day, both at home and at work, and when I hear them now it makes me very nostaligic. The same with the sound of the foghorns (shut down back in the 80's) echoing in from the bay late on a damp and foggy night.

Oh, man, that sounds awesome!!!! I am jealous! I would love to have the experience of living there.

The street cars are so cool! Last time I was there I got to ride on one of the oldschool trolly cars you mention (I think it was called the F-Line along the Embarcadaro).

San Francisco is an amazing place to visit! So much history and diversity. I love all the old buildings/sights/sounds. :)

I would love to capture audio of the fog horns! That would make for some awesome binaural audio. :)

I wish my hometown had interesting stuff like that to record... Oh, well, I guess the grass is always greener on the other side. :(

Thanks again for the reply! Have a great day.

Cheers
Micky