6

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

As the other comment said. Try a low-pass parametric filter. Adjust the frequency down to about 300 Hz, maybe lower, and see how you go.


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

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

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

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

It is impossible to repair clipped signal, since when clipping occurs, part of the original signal is eliminated and cannot be restored. The phenomena is described in the below image: However there are few commercial clicks\pops removal apps available which could improve the signal. The following list of software appear in Wikipedia: Sony Sound Forge ...


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

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

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

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

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

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

There are tons of apps for this. From wikipedia: AthTek Free Voice Changer, Skype Voice Changer, AV Voice Changer Software Diamond 7.0, MorphVOX of Screaming Bee INC, Fake Voice Or maybe try Avox Mutator.


2

You can't actually remove clipping - in recording your levels were too high so your waveforms were damaged at that point. Nothing can bring back the originals, andv even filtering can only be used to remove some of the frequencies that may have been caused.


2

I'm not 100% sure I understand your question, but here's what I think you need to do: Add some compression. This will bring down the peaks closer to the average level. It will only affect the peaks, not the rest of the audio. Normalize. This will raise the level of the the whole audio clip to the specified level. The end result will be less variation in ...


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

This is largely an arrangement question vs a composition question as you already have the material recorded. You need to put on your arranger hat. Besides the cross fade solution I suggest you 'sandbox' the following techniques and then decide which ones work best with each clip and then proceed to the mixing stage. Assuming your audio editor will allow you ...


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

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. ...


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

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

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

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 ...


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