keeping it tight -

a few ideas -

MIDIites out there will know how accurate you can make the timing of a track, but accurate timing in an audio track is more complicated -

FIRST - insert your best song and try this -

Left click on the track speed text next to the green two headed arrow on the righthand side of the transport toolbar, set speed to 0.5x (half forward speed) hit play and listen to the timing of your track, at half speed timing errors will show up, but while track is playing VERY SLOWLY drag arrow to the left so track speed text shows about 0.31x, timing errors will now stand out like a sore thumb -

to adjust timing in a song normally you move a tracks position but when timing is due to irregular drift in the artists performance lining up these timing variations within a track can be problamatic so use SPLICE to cut out small section of track, then you can re-align the drift section without affecting the parts where the artists timing is correct - when timing is correct double click on arrow and normal playback speed will be resumed and song should now sound much tighter -

WHILE IN SLOW MODE do this - open main output VU meter and extend to full height - right click in meter and set to DETAILED SCALE and then set RANGE to 30db - this will help greatly when observing how such things such as playing over joins and fades in tracks effects output (high peaks) that normaly pass to quick for meter to register at normal speed - (LEFT CLICK on large grey button next to track speed text and set to PITCH FOLLOWS SPEED this will show up any BASS overloading that may occur which does not show up at normal playback speed -

the coloured sections in the VU meter follow closely to Bob Katz Katz K12 metering system - what does this mean ? -

the underlying body (average level) of a song should hover closely to the top of the green section, (unless silence is required average level should not drop below -20db, -15bd if possible) - normal changes in instrument and vocal levels will occur in the orange section - parts of the song that need to stand out are allowed to go into the RED section up to (BUT NEVER OVER) -3db - the output level of the finished song may then be increased to 0db (if required) during mastering -

keeping the AVERAGE track level between -12 and -15/-20db makes it easier to setup a track/buss compressor as the compressor range is kept fairly constant and does not bring up the noise floor or crush the peaks as great swings in output level can do -

if you have not used SPLICE yet you will find it under NON DESTRUCTIVE in the EDIT menu - there are 4 options, test them all on a single track to see which suits you (do not save results) -

for more information on Bob Katz metering system go here -

Dr J

This is an interesting idea - I will have to give this a try!

Does that mean I’ll have to eat my words where in the past I’ve called my timing errors “musical genius”? :laugh:

now you can become an genius in “Bio Cordinated Quantization” -

Dr J

Well, you don’t actually have to change playback settings to do the splicing. Simply zoom in close enough to see individual hits in a track. Then splice the track at the points where the hits don’t line up. If you don’t have a midi track of some sort, use the drums as the go-by. They will probobly be the most in-line. Just line up the peaks. This is also useful if you do multiple takes. Record several takes of a lead track (especially good on vocals). Take the best parts of each section of track and combine them in a new track. Sometimes you need to crossfade, etc., but it can really boost up your recording. By the way, remember to try to splice at a point in which the waveform crosses the center line if you can. This will minimize any popping or clicking that can occur when splicing tracks. I also take out as much blank recording as possible. If you have set up your gain balance correctly, then you shouldn’t have any (noticable) drop in background hiss, etc., when you do this. It will also help splicing and prevent pops/clicks.


sure you can zoom in and do it that way, i used to do that and i use volume evolution for track joins etc -

to digress somewhat - recently bought a program called HARBAL which produces an overall frequency graph for a wavefile - it allows you to then make EQ adjustments against a known quantity, IE a peak in the graph that should not be there or a dwell that needs raising up - i bought an RME hammerfall soundcard with the Kats K metering system, and although i find Katz overbearing, his metering system is OK -

although disparate, both items have one thing in common that being ‘frequency’ and to a degree loudness, and that involves knowing how our ears function, when you know the frequency range and the peaks and dwell within the of human hearing and relating that against a frequency graph you clearly see why tracks sound the way they do -

now - from what i have found out it sems that our ears are rather slow at actually hearing and we are not hearing the whole story, so when you slow things down it gives the brain more ‘sampling’ time and allows it to procide a cleaner aural picture, there are things that show up when playback is slow that are just not noticed at normal speed, not just timing errors -

as you mentioned drums you can listen to a drum part at normal speed and it sounds OK, when run slow, you can detect weak beats (where a REAL drummer does not hit the drum with the same strength everytime when they should) - these can be remied in most cases with a slight increase in volume -

with the advent of the DAW another not well understood or documanted effect became apparent, that being in the digital world everything you look at on your screen (this includes metering) is in LINEAR FORM - our ears are not linear devices nor do they respond to frequency as they are displayed on a screen, in other terms what you are looking at can only be used as an indicator and not as gospel, before the DAW there was only one judge and that was the ear, nowadays there are two judges one the ears and the other the eyes, and one sees what the other does not hear - this is one reason why pre DAW recordings sounded better, there was no confusion there was only one judge - and the main reason why so many pro studios do not use a DAW -

the effect of slowing the audio down is to remove the visual aspect from the equasion, OK so its still there but it no longer plays a role because at slow speed it is meaningless, taking the second judge out of the equasion brings hearing to the fore -

Dr J