Normailising, boosting and resampling your audioNoooo! It's not normal!!!You have your WAV file... but it's a little bit too quiet? Here's what you need to do. Method 1: dBpowerAMP Music Converter You may well have this program already due to this guide so you might as well use it. Right click your WAV, select "Convert to" and then choose wav. Set the same settings as your input but select the "Normalize" box. Convert it.
Method 2: Normalize CLI/GUI (better method) Download Normalize and the Normalize GUI Extract and run the normalize gui exe file. You will get something like this: Enter your input WAV, select "Output PCM WAV File" and choose an output WAV. You can normalise to 100% if you like but I prefer 99% to allow for slight accidental gain I might place on the audio in Premiere (I'm daft like that sometimes ^_^) Click Normalise. There you go - nice loud audio without quantization errors. [N.B. If your audio already is oversampled and has garbage data, no amount of normalisation can help you]
Still too Quiet in some parts? Give it a boost...The problem with normalisation is that sometimes the loudest part of your song may happen in only one section. Also, if there are very small clicks and pops in the waveform, they might be the loudest part even if they only happen momentarily. Because of this, the normalisation process may not be as loud as you really want it to be. The solution? Boost the audio by reducing the dynamic range. It makes the quiet parts louder and the loud parts quieter but be warned that you should really just experiment with boosting as it can make your song sound bad but on the most part it is a good method of getting your song to sound generally 'louder'. Boosting your audio using the Boost GUI Go here and download Wavebooster. If you already have BeSweet from an earlier guide, BeSweet also supports boost functions. Unpack and open the BCLI_GUI.exe... Select your Input wav, specify the output wav and it should look like this: |
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As you can see here, the settings I've chosen are Auto, Average dB -6, Maximum dB -.05 which are the defaults for the Auto setting. What the program will do is analyse the audio, and attempt to boost the audio so that the maximum is 0.5 decibels below the max and the average volume of the track is 6 decibels below that. For your particular song, however, you may well need to test different settings. The Maximum value should be fine but if you find it's being boosted too much then make the average dB a larger negative number like -10 etc. Again, it doesn't take very long so it's very easy to quickly test various settings. |
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Changing the Bits and Sample rate of your AudioBits - So, you have an 8bit audio file. Oh well, nobody's perfect. The only problem is with 8bit wav files is that if you want to change them in any way, such as increase the bass or whatever then you will be introducing a second pass of rounding errors. This isn't a good thing really, so before you do anything with your 8bit wav file, convert it into 16bit. It won't make any difference to the way it sounds but it will mean that any changes you do to the file from now on can be done with greater precision due to the extra bits. To change the bitrate, simply use dBpoweramp Music Converter (as demonstrated in the Internet Audio Guide) and select the necessary options in converting to wav. Samplerate - OK, here's the long and short of it. Don't change your Samplerate unless you really really have to. It's a waste of space - you will gain next to nothing by taking a 22kHz wav and converting it into 44.1kHz. All that will happen is that the program doing the conversion will attempt to interpolate new samples in-between the old ones but really it will sound pretty much the same as your original... so don't bother. However, it may well be that your editing software only allows a certain sample rate or you have a 48kHz file that you want to make 44.1kHz so that even really old soundcards can play it. There may well be a good reason for needing to convert your sample rate. So, if you really need to then the best way of doing this is with SSRC. Using BeSweet and SSRC for Sample rate changing:I've talked about BeSweet before in my DVD sourcing guide so for basic functionality you might want to have a quick read of that first. You can get also get the SSRC CLI but I can't imagine many people wanting to type things in the command line ;p Anyway, load up the BeSweet GUI and select your input and output files.. Also choose Wave-Stereo and select the SSRC tab. It should all look a little something like this: OK, now click the SSRC button on the right. Select "Set Sampling rate..." and put in your new sampling rate. Go back to the BeSweet main options page and click WAV to WAV (or MP3 to WAV or whatever depending on your source). That's it, one wav file with a new sample rate. AbsoluteDestiny - June 2002
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