Hot answers tagged audacity
10
I've been using CallGraph to record a weekly podcast over Skype for three years and highly recommend it. It records as either an MP3 or WAV file. http://scribie.com/free-skype-recorder
I use Audacity to edit, but not to record from Skype - it's not the right tool for the job.
6
Vuvuzelas are easily filterable because they are using is a specific frequency. Applause are basically random "clicks" which use a lot of the spectrum, and as such can't really be filterable. You could possibly detect applause and automatically lower the volume. But it's probably not trivial.
You might be able to get a good enough result by using a ...
6
There are a few pretty simple LADSPA reverb plugins that will work with Audacity. Start looking at Steve Harris' docs page.
I like the Plate Reverb he lists for vocals and general reverbiness, but if you are interested in a guitar reverb you want a "spring reverb" sound.
The LADSPA Impulse Convolver has what appear to be compiled-in presets for about 20 ...
5
The basic idea
The characteristic "wobble" bass popular in mid-late 2000s dubstep is frequently (but not always) created by sweeping a lowpass filter over a harmonically-rich sound of some kind, usually some oscillators. The filter cuts out the higher harmonics, creating an effect similar to closing your lips while saying "ahhhhh. By moving the filter ...
5
It is definitely possible that the quality is degraded. All parts of the recording chain have some upper limit on the frequency range they can handle properly. On high-end turntables, this may well exceed the audible range by quite a lot and therefore not matter, but if it was digitalised at 44.1 kHz (which it almost certainly was, if it came back as a ...
4
Since FLAC is the "free lossless audio codec" it should retain the same bit-for-bit audio data. This also implies that the audio parameters (bit rate, sample rate, channel count will be the same. What's not certain is if other meta-data in the .wav file (e.g. comments, cues, etc) will be preserved when converted to flac and back again. If you have none of ...
4
There is a solution to your problem that doesn't involve buying anything. You simply make a multitrack recording, and compensate for latency after the fact by moving your beat forward in time.
Here is an outline of the procedure:
Open up your beat file.
press record, this should record your new material to a new mono track below your beat Press play, note ...
3
While you could create a wobble in Audacity, you might go insane trying.
Instead, download pure data, and work through the tutorials. You don't need to get into much advanced stuff, just learn how to create an OSC~ and a DAC~ and a bit on filtering and FM. Then you can hook up as many oscillators as you want and modulate them and the filters. That's pretty ...
3
To quantify this problem, you want to know the difference between your recording level ("how high the meters are when you're making sound into your mic") and the noise level ("how loud things are when you aren't making noise"). In simple terms, the difference between these is called signal/noise ratio, or SNR. When you boost the signal, you are also boosting ...
3
I'd look for a VST plugin with a water preset. Several of them are discussed here: http://www.kvraudio.com/forum/viewtopic.php?t=174619
I think several effects are required reverb, pitch sweeps, etc.
Some generic underwater effects recommendations here: http://www.musicofsound.co.nz/blog/underwater
3
Since your profile says you are a programmer, then you can fix this fairly easily by writing a small program that will open your WAV file, then copy n bytes, skip n bytes, etc until the end of the file. You can also prove that the repeated chunks are bit identical too.
Off the top of my head, here's the C# code I would write using my own library of audio ...
3
Latency is not an issue of the particular software, but of the sound card drivers. An ASIO driver means the audio goes straight to the sound card instead of going through the operating system. In windows, for example, I've managed to reduce my latency from 44ms using the normal driver, to something like 5ms. The downside is, the operating system can't do a ...
3
As in the answer above this is called a Reverb effect.
I have written some steps below in regards to replicating the effect from the song you posted:
1) Open Audacity, click File -> Open -> Locate the file that you want to apply the effect to
2) Press Ctrl + A to select all
3) Click on Effect -> Gverb
4) Tweak the settings and click on preview to hear what ...
2
If you're using ASIO already you're pretty much at the limit of Ableton Live's (current) ability to handle latency.
However, there are a couple of things you can try. Firstly, you could try to lower the buffer size of your card. This can help a lot, but I've never had too much success with this method, you get audible artifacts if you set it too low, and it ...
2
So the problem is (drumroll) the crappy sound manager that came on the laptop. The RealTek software was defaulted to "filter out noise" from the microphone. It, of course, was doing a terrible job of it. I turned that off and all was well I got tons of extra noise from the un-isolated inner workings. But at least I know where all that sound is coming from.
...
2
In this kind of scenario, where you don't have a multichannel audio interface (or native multichannel functionality) to route Skype through (or the ability to record you and your callee's audio independently) I would perhaps recommend you give Virtual Audio Cable a shot. It costs 30 USD but can give you the kind of flexibility you might be after, ...
2
I've never had much luck removing such things completely, however it's generally possible to improve it with some equalization. Ambient noise, particularly humming or buzzing sounds are often most-annoying at a small frequency range, so it's possible to filter that range out using a notch filter. Generally you can't "magically" get rid of the problem, but ...
2
I'm not sure if you already knew this, but the best way to use GVerb is to duplicate the track you want reverb on and then apply GVerb to one of them. Then you can adjust the level of reverb in the mix.
See: Audacity Wiki
2
You can certainly drop the pitch, but the tricky bit is that a tenor voice has a very different timbre - there are different harmonics, different resonances etc.
The preferred solution would be to get someone with a deep voice to help you :-)
If you have no-one else to help then try a combination of lowering the pitch, and stretching the time frame. The ...
2
In addition to Bart's advice, you can highlight the track and section you want to change, and select 'effects>amplify' to alter just one part of one track.
If the tracks have roughly the same amount of compression, the default offered, which is the highest gain that would result in no clipping, will do a fair job of matching the apparent volume between ...
2
The option you are probably looking for is called Gain. I don't have Audacity myself, but according to the online help you can adjust the gain of each track by the slider on the left of the track. The slider you need to adjust is the one with the + and the - sign below the Mute and Solo button.
In your case you should move the slider of the top track (the ...
2
Never expect to get usable quality when recording through laptop built-in sound hardware, be it the microphone or just the preamp and AD.
A microphone that you can operate at all on your sound card's mic input is almost certainly not suited for piano recording; virtually all such microphones require 48V phantom power.
But you say this problem doesn't occur ...
1
How recently has your piano been tuned? The higher notes have two or three strings per note, so it is possible that the high-frequency phasing is because some of those strings are out of tune by a few cents.
Two other things to consider:
Are you using a single microphone, or either a pair or a "stereo" mic? If you are recording in stereo, try moving the ...
1
This sounds like a phase effect rather than a wah effect. My first though is that you are getting some interaction between the "direct sound" of the piano and reflected sound. I think this is called "comb filtering" and in this particular case is undesirable.
Experiment by moving your microphone position. Also look into "undesirable phase problem piano ...
1
The one thing Garageband gives you in your use case is:
Ease of use!
It is very user friendly for all your basic tasks, and offers you some nice additional features, like backing and drums.
Audacity is cross-platform and for the basics is not too difficult to use, but for anything more than the basics can be tricky to configure, and it sounds like you ...
1
Although Garageband is definitely good value for it's price if you just want to record something you can just as well use audacity.
Garageband has a large library with sound effects and good sounding instruments like drums, strings,trumpets etc. so it's a good (and cheap) place to start with if you want to compose something more than just your voice and ...
1
We have a few other questions on removing a particular piece of audio - the problem is that it isn't possible without damaging the rest of the sound.
If the audio you are trying to remove is very quiet, so you only hear it during quiet parts of the audio, a noise gate will help you, as @Abdias says. If you can hear it even in the loud sections you will not ...
1
This will depend a bit on exactly what is in the files. They may be just audio, but they may have midi bits and pieces in there as well.
In your .band file, you should have a media subfolder, and the audio tracks should be in there (as .aiff files if I recall). Any audio app should let you import them and let you save as wav, mp3 or whatever. I'd use ...
1
Imagine a swimming pool with no ripples. Then drop a pebble in. Ripples will move out from that point. If you choose a point to measure the height you will see it moves up and down from its rest point. Drop another pebble somewhere else and the ripples will interfere, adding and subtracting from each other. The waveform will look complex but the height at ...
1
Sound is a series of negative and positive energy. Technically the Y - Axis does represent "loudness" though loudness can be measured in many different ways. A full positive to negative change is known as a "cycle". Faster cycles means higher pitches, slower cycles means lower pitches. You can simulate this by dragging an object over a textured surface, ...
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