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12

A webcam mic is going to be nowhere near good enough to record infrasound. Most mics, good or bad, aim to capture sound between 20Hz & 20KHz [the approximate absolute limits of human hearing]. The cynic in me would say that roughly, the higher the price, the closer to achieving that they may get. Then there are specialist mics - made by some very ...


8

What you have is too much bass bouncing off the walls. You've created what are called standing waves in your room, which is very common. When there is a standing wave for any frequency, some spots in the room will have almost none of that frequency audible and other spots will have almost double. When you move around, you pass through both kinds of places. ...


7

Yes, you are absolutely right. As a matter of fact, in many simple audio editing programs (my personal favourites are Logic and Audacity, depending whether I'm using my Mac or my Ubuntu), it is very simple to draw a simple wave, zoom in, and grab and move the individual points of the wave to the point which you want. Because the sound is indeed totally ...


6

All of that is perfectly normal. Boost vs. cut What our ears do is not really a Fourier transform like in spectrum analysers1, it's more comparable to an auto-correlation function. For instance, we still recognise pitches without a problem even when the fundamental frequency is completely missing! This is useful in natural hearing environments, since ...


6

It is effectively a sample, but the comparison isn't quite exact. Photos have measurements of intensity on two axis with each pixel having an intensity measurement for red, green and blue. A sound sample is the measure of the intensity of an audio signal at a moment in time, so it is kind of like a cross between a pixel and a frame. There are generally ...


6

XLR typically gives you a 'balanced' connection, whereas the Jack connection is unbalanced. An unbalanced connection is very susceptible to electromagnetic interference - the mains hum you get from a guitar plugged into an amp, for example. For a high signal level instrument, like a guitar, this is often not a problem, but for a low level signal like a ...


6

This is probably a better question for Sound.SE, but anyway: A good vocalist will be aware of their mic placement (it becomes habit pretty quick). In "loud" genres, rock for example, as an engineer, you typically want the mic very close the vocalist's mouth. The closer the vocalist (and louder the vocalist), the less gain you require, which improves your ...


6

There are three parameters of this filter that are described in the phrase "100 Hz 12 dB per octave low pass filter". I'll cover them in reverse order. Low pass filter - This means the filter does not change lower frequencies ("passes" those frequencies through) and blocks higher frequencies. Sometimes these filters are called "high cut filters", but that ...


5

You can use the amerge and pan filters in ffmpeg to combine two mono streams into one stereo output: ffmpeg -i input -filter_complex "[0:3] [0:4] amerge,pan=stereo:c0=c0:c1=c1" -c:v copy -c:a pcm_s16le output or using -ac instead of pan: ffmpeg -i input -filter_complex "[0:3] [0:4] amerge" -c:v copy -c:a pcm_s16le -ac 2 output [0:3] and [0:4] refer to ...


5

A voice is a very complex sound pattern. While you can manipulate a male voice to sound feminine or vice versa, getting it to sound like a particular person is probably going to be pretty difficult since the patterns of frequencies differ in complex ways. Think about it this way, think about voice synthesis and how we can't even make a computer generate a ...


5

Here's my process for creating loopable sound files: Select a reasonable amount of audio from the tail end of the file ("reasonable" is variable and entirely dependent on the content of what you're working with), cut it off and paste it onto the front of the file. You'll need to make a crossfade from the tail end of the region you just cut onto the head of ...


5

Actually you can do soundproofing without damaging the house. Using acoustic panels (expensive but provides the best result) works, however some well placed blankets using string and 3M adhesive hooks in a carpeted room can vastly change the sound of the recording because of reduced reverberation. Picking a recording time that has the least amount of noise ...


5

It depends on what you want to do. With just FMOD Studio, you can build complex events with parameter-driven effects, detailed internal signal routing, randomised elements, and internal trigger logic of a variety of different flavours. You can also mix your project, devise snapshots and sidechains to alter how it behaves under particular circumstances, and ...


5

I strongly disagree with a lot of Dave's recommendations, based on my experience recording vehicle sounds for racing games, advertisement and sound engineering: an engine under load doesn't sound like an engine revving while standing still at all. "patching in after the fact" will be loads more work than recording audio and video at the same time. Expect ...


5

If you want to save some money and do it yourself you might try suspending a membrane of some sort, like a piece of paper or plastic, and attaching an accelerometer to it. A quick search found the MMA8453Q made by Freescale Semiconductor. It costs a dollar, takes 10bit samples, and samples at a rate of up to 800Hz. The max sensitivity mode is 2g which ...


5

In addition to a little reverb (convolution reverbs are critical tools for sound for picture, IMHO), it will still sound like soundtrack music with extra reverb unless it is filtered. You could run it through a speaker emulator or guitar or bass amp plug-in, or you could do high and low pass filters along with a little high shelf. Instead of low pass + high ...


4

Not really, no. Only the most advanced surround sound systems support proper vertical panning (it would be called tilting I suppose). All standard systems have at all or at least the crucial speakers1 on the same vertical level, so there's not really any elevation information present. Any perceived top-bottom separation has indirect, psycho- or room-...


4

The sound becomes higher in pitch. Pitch is, basically, just frequency. "Shriller" is one way to describe this, although it carries a somewhat negative connotation.


4

What an exciting journey! My two cents worth: Regarding the safety of your data, be sure to upload to the cloud (Dropbox or the like) whenever you have internet. Bring WAY more memory cards than you think you will need, then double that. As long as you can get it to the cloud and know someone back home has copied it locally AND backed it up preferably ...


4

Talking to schizophrenics for advice how it sounds won't necessarily do you much good as we're not talking something like an LSD-trip or such here, we're talking a state that appears totally natural to the senses of the one suffering it, and this differs from person to person, as well as there are rarely dronings or similar sounds present. With rarely I mean ...


4

In general, the creator of a video can choose any combination of video-quality and audio-quality. In practice, persons who know what they are doing, and also the automatic engines of YouTube (facebook should be similar) that create different quality versions of the same video, have profiles, where the quality of video and audio is linked. So yes, higher ...


4

Two thoughts... 1) The song is currently in the wrong key for the sound you are aiming for, vocally. The only fix for that would really be to re-record it in a more suitable key. 2) There are plugins available that can adjust the formant** of a sound without changing the pitch. Probably the best tool available for that is Melodyne [imho] which can adjust ...


4

The only way to soundproof anything is mass. You need weight. Nothing else will work for bass frequencies. No amount of egg boxes will have any effect. Assuming you actually still need access to the door & that rebuilding the entire thing triple-glazed is right out of the question, then the only thing I can think of might be lots of heavy blankets, ...


4

Usually the smaller more intimate sounds will provide the detail, so you'll have those larger, slower, heavier wave sounds as the body and you can provide movement and texture and detail with smaller splashes, sprays etc mixed in.


4

To get the best audio quality in your final lossy-eccoded audio file output at the smallest possible file size, you have to do 3 steps: sample rate convert from 96kHz to 44.1kHz, because the lossy formats do not support 96kHz dither the 24-bit file to 16-bits because the lossy formats do not support 24-bits, and if you don’t dither, all you are doing is ...


4

The channels of the respective output format (2.0 / 5.1 etc) are exported as separate mono audio files. Usually the encoders (DCP / Blu-ray Etc.) expect separate audio tracks as input. Apart from that a cinema mix needs to be mixed in a certain dynamic range similar to ebu r128. Check I.e. this video for an introduction: ...


4

an attempt at a simpler answer... what audacity is showing you is the sample of the whoooole song jammed together in time. every sample for every note (of every instrument) has already been mixed down into ONE wave. This is the wave your speaker plays out. So showing music this way is a terrible idea. A human can't separate out which note of which ...


4

The highly simplified answer is that sound in the real world is not sine waves with fixed frequencies. We learn about sound initially as frequencies or notes and simplify the concept down to thinking about various sine waves to make it easier to understand the fundamentals of what is going on. The reality, however, is that all sound propagates through the ...


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