7

That's a well-known problem. First of all, you shouldn't be using the microphone connector on the laptop. These inputs are usually not only particularly susceptible to buzzing noises, but also to all kinds of distortion and aliasing. Use an external audio interface, there are very affordable USB ones available. Using an interface with XLR inputs, connected ...


7

One way that I've used it in the past is to create a side-chain gate on an Aux. Feed your original signal through a bus to that channel, and place your distortion of choice after the gate in your plug-in chain. This way you can control when the distortion kicks in against the original's amplitude, and you have additional wet/dry mix control through volume ...


7

Nope, the sound information is missing and there is no way to recover it. (At least not that I'm aware of.) Even if there was, it would have to basically be completely guessing at what should be there. (Edit: there is software that will make the guess, and that's what the answer with the waveforms is illustrating. It is worth noting that it is a guess ...


5

Nothing posted by Schizomorph is wrong - if it's busted, it's busted... However, one small ray of light is azimuth/stereo balance correction in software. Back in the old days I had a bit of freeware (and the source code) by Airwindows - one of the most amazing, if just a little esoteric, sound designers on the planet - that could do this task [non-realtime],...


5

I may be wrong but this sounds to me like a simple ring modulator.


4

I'm afraid you can't re-construct information (in the waveform) that has been distorted. The one side you can improve is playback quality by using a decent tape player with a good head. Tape distortion flattens the peaks, introducing odd harmonics. If there was only one frequency, you could filter those out but since it's a complex waveform we're talking ...


3

Well I've tried to come close : I used reaper, Izotope Trash 2 for saturation and Izotope Ozone 5 for multiband compression. 1. pitch shift - 3 semitones 2. Cut some low freqs with EQ 3. Izotope Trash with these settings : http://i.imgur.com/Sd5k2lq.jpg 4. Izotope Ozone with these settings : http://imgur.com/a/6RC4G You ...


3

What you are actually hearing is square waves. What happens when a speaker is fed a signal that exceeds the amplitude it can produce is that the speaker coil projects the diaphram to the limit of how far it can travel. This results in an abrupt stop to the sound pressure and it stays pegged at the extreme end until the signal amplitude falls enough for it ...


3

I usually find that SM57s accentuate distortion/tinniness when the mic is pointed at the center of the speaker cone. My preferred placement is 45 degrees off-axis, aimed between the outer edge of the cone and the outer edge of the speaker. Tends to give a fuller reflection of the "sound in the room"... I'm not sure that this would totally mitigate your ...


2

I love distortion and use it an a lot of things, but it is, as you said yourself, easy to over-do it. It's not easy to put in words how I work with it as it would need a common reference of some sort to make sense, but if we compare it with spices, all sorts of creativity are based in the same way but with different casings, you must keep a balance in the ...


2

There's of course plenty of ways to do that. The easiest way on Windows would certainly be to load both a SoundFont synth and an amp-simulator plugin into a DAW, but I suppose that's not the solution you want. If you'd like to keep it down to a single executable and, as I suspect, you're more interested seeing in how intonation influences distortion-...


2

Unfortunately it is tough to generate a good sounding distorted guitar using only digital means. Although as sampling gets better this is slowly getting better as well. Your idea of sampling a clean guitar sound and then routing it though a distortion unit and back into your DAW would be the route I would take assuming you cant get your hands on a real ...


2

Check this Wikipedia article The article doesn't go into full detail, but let's address yo your problem: YES Audio loses quality very easily specially when changing its sampling rate and in your particular case it probably has to do with aliasing, which is a consequence of re sampling audio. Basically - as people tell me at least - you should only raise ...


2

Try Speakerphone from Audio Ease. It has a dropped frame simulation you can hear starting from 15:20 in their Youtube video on this product page http://www.audioease.com/Pages/Speakerphone/speakerphone.html


2

I think it would be better to find other sound effects... Or, better yet, make them yourself. Cleaning those sounds will probably be a lot of work for a non ideal result. Additionally those sounds may be protected by author´s rights and you could have legal problems using them. Enjoy the process of your sound effects creation!


2

Ohmicide from Ohm Force is specifically made to provide over-the-top distortion effects. Available as 32/64-bit Audio Unit, VST, and RTAS.


2

I have just seen the following article where this sound effect is mentioned (http://www.latimes.com/entertainment/envelope/la-en-craft-star-wars-sound-20160204-story.html). It is interesting to note that the dialogue was recorded with two mics: one in the helmet itself and the other on the chest. Clearly the timbre of the two recordings will be different, ...


2

I agree with Mark. A Tape Emulation plugin might be a good option. I can highly recommend Satin by U-He. it has great options to beautifully (or badly) degrade sound (besides enhancing it nicely). You can add some distortions (try a slight amount bit crusher) - lots of free plugins out there. simply google it for your desired platform and DAW.


2

What you are asking for is pretty much like trying to take the sugar of a baked cake. There are tools that could help but I have no personal experience with them. From what I was told, iZotope RX can work some magic. I think they have a trial version, but I don't know how limited it is. Using a software like audacity or reaper you can also put a hi-pass ...


2

No, I don't think you can fix that unless you have a fully equipped studio for forensic analysis, very good ears and a LOT of time. What you're hearing has a number of problems. The greatest is that its time is messed up due to a (circular) buffer that wasn't filling fast enough. This is what causes these repetitions. If I can hear well, a second track (the ...


1

XLR inputs are mic inputs on the H4n, not line. Put a 40db pad or so on the line and it will probably work ok, though you may still have some quality loss from impedance differences depending on the output.


1

32 ohm is a moderately low impedance and fairly easily driveable if your soundcard has a moderately powerful amp. Sounds like it's capable of overdriving the cans, and the drivers aren't able to gracefully handle the high level input without clipping. Do they behave when paired with a matched preamp? Is their frequency response ok?


1

Quick analysis: It has a cycle running through it, like a cricket's chirp. Maybe tremolo, but probably part of the wave shaping process- a cycled wave, maybe originally a much lower pitched looping sound. Overall distortion effect sounds like a subtle Bitcrusher. It's very wide, spatially. Envelope has a Fast decay, low sustain, no release. Everything is ...


1

Perhaps not the most practical and flexible solution: you don't have to emulate, you can apply manual changes in bandwidth using a tool like NetBalancer and you can start some heavy downloads, use a speed tester etc to strain the connection


1

mm that's a tricky one , well lets think about the distortion a bit,the creamy top end makes me look at a fuzz, maybe an octa/fuzz with the octaver low in the mix, cause i hear some low end which i cant really explain otherwise (but that does not refer to the core of the sound). So , yea, Fuzz would be my go to effect in that situation (it's not a very high ...


1

I think this has something to do with HARMONY, i haven't completed harmony yet so I'm not 100% sure... When you have two frequencies of different values playing simultaneously, the brain and ear automatically begin to compare the two intervals. Unless it's by like a decimal it will be noticeably different than the first.


1

NO, resampling – provided it's done properly – does not degrade the quality. In any sampled audio signal, the assumption is that, prior to sampling, it was ensured that the signal only contains frequencies up to the Nyqvist frequency before sampling; for this purpose AD converters contain low pass filters, which don't work perfectly but can work very good, ...


1

The question about the nature of clipping phenomena was discussed many times here on this site (for example here) and on dozens of other places which are easily googled. So as it was already discussed, clipping - is just few points where signal amplitude exceed the available range of finite amplitude levels. This causing the wave signal to become less ...


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