Audio Levels

[freeway]

Hi John

Not sure if I'm in the right thread here apologies if so.

As you might have noticed from MH I recently changed over from using mini disks to using an Ipad . Since I've changed over the volumes in some of my files seem to louder or lower than others . I never had that problem before and all levels were fairly accurate .

I use converted midi files and real audio files and I notice a big difference in the levels on both. The audio files being much louder and fuller prob because of the real instrumentation as opposed to midi.

I've tried MP3 Gain and have them all set at the same level I play them also through my mini disk using the recorder levels to make sure that they're not too low or dont peak before I transfer on to the iPad but when I gig I notice a huge difference in volumes.

Is there anyway possible of having them all set at the same level as its very annoying and time consuming reaching over to my desk every 5 mins.

I tried bringing down the levels of the audio tracks to compensate but their still either too low or too loud.

Any help greatly appreciated

Rgds

Barry

[JohnG]

Hi Barry,

No, you have it right, this is the section for questions not directly related to any specific tutorial.

First let me be quite honest, I know next to nothing about iPads or any other tablet, and it's highly unlikely that the situation will change anytime in the near future. I can see though, that they could be a godsend for a gigging musician.

The fact that some are made from MIDI files shouldn't make any difference but, of course, it will depend upon the sound module and effects unit being used. The only thing that I do with all my files, unless I deliberately want them softer, is to process every wav file through Audacity setting the peak level to -3dB, before conversion to mp3.

I take a look at the levels of various sections e.g. verse compared to chorus, to make sure the comparative levels are right. I then compare what I've done with a reference file to see whether they sound about the same in volume. Sometimes a very loud section in one file needs to be lowered a bit to bring in order to be able to bring the overall level up a bit.

Often just pulling the audio peak level down doesn't sound correct, so I go back to the MIDI file and often find some people use note on velocities way up at 127 or near. Pulling all these down a bit to sensible levels helps significantly. It's a lot of work though going back through all the files.

I'm not sure how much that helps, and I'm afraid I don't have a quick fix to offer. I suspect that the conversion to mini disk in the past, just brought everything to the same level automatically.

Regards,
JohnG.

[freeway]

Hi John

Thanks for the speedy response yes the iPad is a godsend for me especially not having to lug disks and books of lyrics around anymore. I also find I can now concentrate fully on my audience and setlist without the need to be turnng around and fumbling through disks and lyrics.

Anyway about the volume levels I might give Audacity a try at the levels you suggest . I'm not really sure if Wav files have the edge over Mp3's soundwise or not but its mostly the latter I use and I find them pretty good its more a matter of convenience than anything else that I choose Mp3 as space (16GB ) is not an issue for me .

So I presume it really doesen't matter either Wav or Mp3 soundwise they should be more or less the same I'll give Audacity a try as this is bugging me no end and even if it take me a bit of time and effort it'll be worth it in the long run.

Just in Audacity there ....is it the Nomalise function I use to set the levels at-3db are there any faders as such for on the fly changing? I'm relatively new to Audacity so is it then just a matter of recording the song at the chosen level and then exporting when finished?

Appreciate your time I'll let you know how it goes.

Thanks again

Rgds

Barry

[JohnG]

Hi Barry,

Okay, quick tutorial.
I tend to use Audacity in full screen view.
Open the wav file and once you have the waveform displayed Alt then V then V gives a magnified display.
Ctrl and A at the same time selects all (as in e.g. Word).
Then in the menu "effect" and "amplify".

Now in the top box you'll see how much you need to amplify in order to reach 0dB (this is too high a level for conversion to mp3).
In this box you can type +ve or -ve numbers e.g. -2.3 or 2.3 giving a reduction or amplification respectively.
The box below shows the resultant maximum level should you apply the change. I always leave 3 dB headroom for the conversion to mp3 process.
I set the overall level first.
But you may want some parts e.g. one verse reduced and others parts e.g. a chorus increased.

You need to select the bit you want to change by using the "I" mouse pointer (the default) to highlight the selection. Drag from one end to the other of the bit you want to change. It becomes highlighted.
It's best to use the magnifying glass (there's a set at the top right tool bar) to "Zoom In" to a good level before selection. It's possible to zoom right in to sample level, but usually not necessary unless you have clipping, then you can edit down the clipped samples.

Once you've selected the part you want to alter you can then change its level, again using amplify.
Providing the edits aren't too drastic the result will hang together remarkably well.

You can edit out or insert silences at beginning or end (or anywhere).
When I've done I then save the edited wav file with a new name using the export function.
Then I export again but this time to mp3 (you need to install the LAME converter the first time).
I think the audacity web site points you to the right place for this.
I use the preset "extreme" and, to be honest, the result is indistinguishable to the wav for me.
But, past retirement age, the hearing is not what it was 50 years ago. ;-)

Does that help?
Regards,
JohnG.

[freeway]

Many Thanks for that John

Getting a small bit confused about setting the dbs would you mind elaborating... enclosing a link to a pic tried to insert pic but thats for another day.....anyway after loading an mp3 where do I go from here kinda stumped ?

Appreciate your time

Barry

www.box.com/s/w3s215pt5a95v0cvdcii

[JohnG]

Hi Barry,

What the amplify box is telling you is that the file is currently 2.6dB below peak level.
Or, as near as makes no difference, to 3dB below peak.
It is showing that 2.6dB of amplification (top box) would make it 0dB (bottom box), peak digital level.

With digital signals we always use zero to indicate the maximum level that can be recorded using 16 bit samples. Try to go over that and the top of the signal get s chopped off. This is different to analogue where we often get many dB of "headroom".

So with digital we either write e.g. 2.6dB below full scale deflection, or -2.6dB FSD.
The FSD is often left off when we know we are talking digital audio.

I think the reason the picture won't display is because Box.com doesn't host pictures in the same way as e.g. photobucket. Most of mine are hosted there. I tried putting img round it but no go.

The file you're looking at looks about right. It probably started out as a -3dB FSD wav file and ended up -2.6dB FSD after conversion to mp3, at a guess.

Is this a problem file?

Just thinking, I wonder if your program treats wavs and mp3s in a different way?
Is there, I wonder, a way to change the settings if so?
Just thinking aloud.

Regards,
John.

[freeway]

John

Thats great I'll give that a bash over the next few days hopefully this will sort out the volume issue once and for all.

Once again thanks for your time and patience.

Rgds

Barry

[JohnG]

No problem, Barry.

Let me know how you get on, would you?
As an afterthought, I wonder if it would be better if all the files were in mp3 format.
The wav files should lose very little providing you use a good converter (I always use LAME) set to a higher sample rate, minimum 192 kbps, but better 256k or more. I always use 256 or even 320 kbps.
Most files do lose something at 128k.
And some mp3 converters are definitely better than others.

A good general purpose one is RazorLame.
It too uses the LAME converter, providing an easy to use Windows interface.

Regards,
JohnG.

[freeway]

John

Thanks for that tip I usually left the sample rate alone at 192 didn't realise it'd make much of a difference now I do

I suppose also the original track has to be of good quality to start off with otherwise the more it detriorates with conversion the worse the finished product will sound .

All my files are mp3s so don't have anyworries with wav files ..I'll let you know how it goes for sure.

I'm also messing about with Mixcraft which is also excellent and I'm more familiar with that Prog so I'll find a happy medium between the two. Once it does the job for me I'm happy out .

Thanks again

Barry

[JohnG]

Hi Barry,

I'm sure you realise that there's no point converting an mp3 back to wav then reconverting to a higher rate. Once the more detailed information has been discarded in the first conversion it can never be recovered. The second conversion will likely throw away a bit more. But I'm sure you knew that. It's only worthwhile during the first conversion process from wav to a higher bit rate mp3.

Yes, of course, if the original wav file isn't up to scratch then you'll just end up with a compressed version of the same. At reasonably high conversion rates, 256k or more, the degradation should be minimal, hardly noticeable with a good converter. Again I'd recommend the LAME converter.

The program RazorLame is one I'd recommend. The results can be very good, ask our friend Simmo, he uses it to create his backing tracks.

The key to getting good files is to work in the wav domain for as long as possible once a good MIDI file has been created. Only when you are totally satisfied with the result do the final conversion.

All the best and 'my pleasure'.
Have fun.
JohnG.