7

According to my copy of the absolutely essential (!) Master Handbook of Acoustics, to hear shorter tones, that is sounds with a short impulse, they need to be louder: A 1,000-Hz tone sounds like 1,000 Hz in a 1-second tone burst, but an extremely short burst > sounds like a click. The duration of such a burst also influences the perceived loudness. ...


6

Is AC current fundamentally better as a working current when it comes to amplifying audio signal? No. If anything, using an AC power supply makes it more difficult to have low noise. The circuits that handle audio are all powered by DC. So any amplifier that has an AC power input has to convert this to DC internally. The AC cabling inside the amplifier ...


5

Virtually all modern audio interfaces for computer use have clock-synchronized sampling across all inputs. This is a prime requirement for most ordinary multi-track recording. It would be difficult to find a modern, multi-track (>2) audio interface that did NOT meet this requirement.


5

I agree with what's been said, although don't forget that part of what gives a sound its off-axis/down-the-hall timbre is how it resonates through the building materials. So yes, highs will drop off but you'll likely need to bump sonewhere between 180 - 400 Hz where there's a nice resonant quality, just be careful of the 300 Hz muddiness. This is where a ...


5

The type of current is actually completely irrelevant as all transmitted power is AC. Unless you are driving your amplifier from batteries, everything has to be converted at some point to DC. The end result all boils down to the amount of power you need to draw from the supply in order to amplify the audio to the level you require. The main reason why AC ...


4

I've had great luck using a home made impulse response created by slamming a 2x4 plank on the floor upstairs with a hammer hard, while recording it downstairs. Try taking the top of your sound off with a filter and send it through an IR like that.


4

This question, as stated, is not answerable. The answer is "it depends". What you need to determine is the noise level in the file. Silence is rarely silent, but there is a large difference between if there was a quiet fan going in the background vs recording in a crowded and noisy room. It will also vary greatly between different recording hardware as ...


3

You need to understand sound sources (oscillators, noise generators, and samples), modulation (LFO in particular), sound shaping (envelopes), and then signal processing (read: math and audio physics). And honestly I'm being really high level here. There are a million details within each thing I mentioned. Its not really difficult, but I think it does ...


3

DJ software like Native Instruments Traktor is pretty good at guessing the tempo by analysing a music track. It can also send out a MIDI clock signal that can be used to sync lights or something to. Ableton Live and some other programs can do Audio-To-Midi, with varying success: it works better on simple soundfiles and much worse on full musical songs with ...


3

The shortest length of time your eardrum can respond to a given sound depends on its frequency. The higher the frequency, the shorter the response time and the shorter time required for the ear to register a perceptible noise. Thus, for one who's highest audible frequency is, say 16kHz, the shortest length of time the sound would have to last is 0.0000625. ...


3

If you create a soundfile at 44100 Hz with only silence in it, and set exactly one sample to a non-zero value, you will hear the non-zero sample as a click. This is essentially a sound of 0,0227 ms (the duration of one sample). You can do this with Audacity, or any wave editor that lets you zoom in onto a WAV-file and move the individual sample points around....


3

There is no shortest sound detectable by the human ear. Any impulse with enough energy is audible. The shortest detectable tone, identifiable as a tone, would be on the order of 100 ms. It might be shorter for tones of high pitch.


3

Another option is to look into Sonic Core's products. Originally, the PCI card-based hardware was developed by CreamWare, but Sonic Core acquired them, and developed the Xite series of hardware based on the original technology. The hardware can host Effects and Instrument (proprietary) plugins, which are actively developed (both commercially and through a ...


3

I would like to expand on the caveat in @Rory Alsop's answer, since to me he answers one part of your question (can I play it back so I can hear it?) while relegating the arguably far less trivial part (can I record an ultra/infrasonic sound?) to a caveat. So what are the hardware/software requirements to record inaudible frequencies? I will concentrate on ...


3

In addition to @frcake's excellent answer, I have a few points: I have been gigging for over 30 years and I still have trouble getting some sounds I want. One solution that most guitarists go for is to have many guitars. All of mine sound slightly different, sustain, wood tone, pickups, bridge, electrics, resonance etc. Some have higher action or greater ...


3

As long as you have identical rolloff characteristics - slope and frequency, the mix of the two will be identical to the original signal. Whether the filter alters phase or not depends entirely on the design of the filter. For instance a basic first order R/C filter will alter the phase by between 45 and 90 degrees depending on slope and frequency. There ...


3

Use the Marantz as your pre-amp. Route the audio out of one of its 'Recorder' outputs. This should give you consumer-level output, nominally -20dB. Input to the Behringer on a switchable input mic/line/instrument & flick between the options to see which gives you the best impedance/level match - probably line or instrument. Watch levels as you do this, ...


2

if you're still interested you might want to check komponant.com We're creating a technology that does exactly what you're looking for Cheers


2

I'm a little late to the party, but I've used ProTools' 7-band EQ plugin for replicating sounds coming from inside of a building, and it sounds great. There are some other things I would do to adjust for specific needs, but this is a good start. Here is a screenshot of the EQ settings: If you were still looking for it, I hope this helped.


2

Alter Ego Bones (free) Standalone, VST % AU 64bit - wi;; sing and speak whatever you type in to it, same presets are provided in English and Japanese, there are also add-on voices which are not free. https://www.plogue.com/downloads.html


2

In one sense, the answer from "audionuma" is correct, a similar process is possible in both Ableton and Audacity. In Ableton, all the tools you need to do this are contained in the "Utility" plug-in, which you can use to phase-invert one recording and thus "subtract" it from the other.... HOWEVER** Neither Audio diff-maker nor the more basic process as ...


2

In the case where your two audio signals S1[n] and S2[n] are of same length, and we are speaking of discrete time, discrete value signals, the DFT being a linear transform : DFT (S1[n] - S2[n]) = DFT(S1[n]) - DFT(S2[n]) It means that subtracting the spectrum of S2[n] from the spectrum of S1[n] and transform back into time domain signal will produce the ...


2

You would rather turn to a program like Matlab or its open source equivalent Octave. Open your mono-channel audio data as a vector (using wavread for instance), and turn each sample value to its square value. Then compute the mean every N samples, N depending on the smoothness/time-precision you need.


2

In the simplest case, Fourier analysis probably won't help understand this. You can think of it arithmetically. (L - R) subtracts the right channel from the left channel, so as you noted, those parts they have in common -- those that are correlated -- will cancel. The amount of correlation determines the amount of cancellation. Now, if you want a much ...


2

Honestly, I think you'd have a much simpler task if you tried some de-clicker software on the best of the pair than attempt to keep two independently-recorded analogue tracks in prefect phase-sync for three minutes whilst cross-cutting by eye. I know of no software that could automate your chosen task. My de-clicker of choice is Waves X-Click Two others I'...


2

A microphone usually measure pressure variation or pressure gradient of the acoustic sound wave at the transducer location. That means that the atmospheric pressure (what you possibly call ambient pressure) does not generate output on a microphone. What you are actually measuring at the output of the microphone is the acoustic wave pressure oscillating ...


2

As a rule of thumb, closer to the center is trebly , closer to the ring is mellower - more bassy. Electric guitar recording is a world of it's own, many people go to insane extends to hunt the tone they want. First you have to fix the tone way before the recording stage, for instance in a song as soft as the one you posted, you can't just have an accented ...


2

The answer is to be found in understanding what a crossfade is. A crossfade can be either "Constant Level" or "Constant Power". One is linear, one is logarithmic in shape. You need to build a table of coefficients (float values between 0 and 1) that represent the shape you require. I would think you need a logarithmic fade for this application. You need ...


2

The process you are using is not going to work, as you have a one-to-one mapping of input to output. Any time you do that, patterns exist that relate to the original. This type of algorithm has been attempted many times in various fields, and it never works to protect/obfuscate. This is why we tell security/crypto/dev folks to never roll their own crypto. ...


1

I don't know if this will satisfy the "dirty" part, but you could use a free (freeware/open source) digital audio editor to get it done quickly and cheaply. I'd suggest using Audacity. It's a popular, and in my experience, capable open source digital audio editor, which, with a little direction, you could quite easily utilize for all your needs. And because ...


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