- Djay 2 Slow Down Without Affecting Pitching
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Introducing djay 2 - the next generation of the #1 DJ app is now available on Android. Djay transforms your Android device into a full–featured DJ system. Seamlessly integrated with your music library, djay gives you direct access to mix your favorite songs and playlists. Apr 12, 2011 Re: slow down audio without pitch change Post by njckid » Tue Apr 12, 2011 10:43 pm i struggled with this issue as well but it is very easy. You click on the clip (has to be audio), you put down two warp markers around what you want to change, then eather click and drag one of. Ready to rock your next party? Introducing djay 2 - the next generation of the #1 DJ app is now available on Android. Djay transforms your Android device into a full–featured DJ system. Seamlessly integrated with your music library, djay gives you direct access to mix your favorite songs and playlists.
The program 'Transcribe!' (of which I am the author) is intended to help musicians totranscribe music from recordings. It has the ability to slow down music (or speed it up) in real timewithout changing the pitch. People sometimes ask how this is done so I have written this discussionof the subject. It is a fairly general discussion but I also comment on the particular methods whichI chose to use in Transcribe!
I mostly discuss slowing down rather than speeding up here, but with a moment's thought you cansee that pretty much every issue discussed applies equally to both.
What Do We Really Want?
If you have never tried it then you might think that once you have some music on your hard diskin digital form, it would be easy to change the speed without changing the pitch - just a bit ofresampling or something like that. But in fact it's difficult. Resampling - changing the sample rate- merely enables you to change the pitch and speed together in a way that's exactly analogousto varying the speed of an analogue tape recorder or vinyl record player. Halve the speed andthe pitch goes down an octave.
Djay 2 Slow Down Without Affecting Pitching
Before we can decide what we should be aiming for we have to ask what our slowdownprogram will be used for. Here are some questions we must consider:
- Are we working on polyphonic material (many notes at a time) or just one note at a time?Different techniques may be preferred, depending.
Djay 2 Slow Down Without Affecting Pitch Youtube
- Are we running in real time on a desktop computer in which case we must use efficientprocessing techniques, or are we running on expensive dedicated hardware or non-realtime,in which case we can use more sophisticated (slower) methods?
- Are we aiming for a high quality realistic musical result, in which case it is reasonable tolimit ourselves to small changes in speed (because large changes can never sound musicallyrealistic anyway) or do we want large changes, in which case we must accept that it will sound'processed'?
- Most instruments have a clearly defined start-up noise at the beginning of each note, whichis quite different from the sound of the sustained note - for instance the tonguing of a brassinstrument, the hammer of a piano, the hit of a drum. If we want the slowed-down music to soundlike a real player on a real instrument who simply happens to be playing more slowly then wemust not stretch these 'transients' but only stretch the sustained notes or silencesbetween them. On the other hand we might want to hear the details of the attack, for instanceif we analysing a player's technique for teaching purposes, in which case we want to stretcheverything equally.
- Suppose we have a sustained bass note at 31Hz (B, the open bottom string of a 5 stringbass). When we slow this down we would clearly want the note to remain at 31Hz (31 cyclesper second) so the slowdown method must generate more 'cycles' of oscillationto fill up the extra time. On the other hand suppose we have a guitar strum where the guitaristsmoothly strums all 6 strings in about a fifth of a second - a moderate speed strum. In thiscase the strum has about 30 note-attacks per second. If we slow this down we want thenote-attacks to be spread further apart while there are still 6 of them. So here we havetwo examples of sonic events which happen at about 30 per second, for which the desiredhandling is radically different. How can any slowdown method know which approach to use,or even separate out the sounds to be handled with different approaches given that the bassnote and the guitar strum could very easily both be happening at the same time?
Most commercial speed or pitch change software is intended for music recording andediting applications - for instance changing the pitch of a singer's voice to correct anout-of-tune note, or adjusting the duration of a music clip to make it fit an advertisement.In that case a natural sound is vital, but there is no need to support changes greater thanabout 20 or 30% either way, as anything more than that stands practically no chance ofsounding natural anyway. High quality programs in this area do indeed make the effort oflocating transients and not modifying them, and many other sophisticated techniques. Forthis to work in real time you would usually be talking about a dedicated DSP processing effects unit.
Transcribe! is intended as an aid for transcribing and needs to slow the music downmuch more drastically - by a factor of up to 20 in fact - but 'natural' sound isfortunately not such a priority, instead the priority is to be able to hear clearly what'shappening. For this reason I regard it as more sensible - and easier - for Transcribe!to stretch everything equally.
The bass note vs. guitar strum example above is a tricky one but multi-resolutionprocessing (see below) helps a lot.
By the way, once you solve the problem of changing speed without changing pitchyou can easily change pitch without changing speed by applying a touch of resamplingafterwards. For instance if you want to raise the pitch then you first lower the speed withoutchanging the pitch, then resample to speed it back up to the original speed while raisingthe pitch too. There are also more direct ways of changing the pitch which I won't be discussing here.
The basic technique used by most slowdown methods whether 'time domain'or 'frequency domain' (see below) is to slice the input sound into short segments- typically in the range from a 100th to a 10th of a second - to spread those segments furtherapart in time, and to fill the gaps by duplicating bits of the segments either side - a sort of'copy and paste' into the gaps. There are also 'modelling' techniqueswhich attempt to analyse the material at a high level and then reconstruct at slower speed fromthe high level description. These can be good on certain material but I won't be discussing them here.
Apparently back in the steam age you could get tape recorders which implemented thistechnique mechanically. The playback head was circular and in fact had four playback headsequally spaced around the circumference. The head rotated while the tape moved past it anda brush contact underneath ensured that the head which was currently in contact with the tapewas the one whose output was fed to the playback amplifier. The overall speed was controlledby the speed of the tape while the pitch was controlled by the relative speed of the tape pastthe playback head which would not be the same if the head was rotating. You can see how thisinvolves playing certain little slices of tape twice as one head takes over from another.
This crude technique (whether implemented mechanically or digitally) is easy to do but hasmany problems with sound quality. The splice points introduce discontinuities in the sound andas there are perhaps 30 splice points per second, this causes a dreadful warbling noise. Alsotransients are duplicated if they happen to be in a segment which gets used twice, creating asmeared-out effect which becomes very bad at high slowdown ratios. The rest of this discussionwill be about some of the techniques we can use to try to reduce these bad effects. The basic ideais, we must analyse the sound to some extent, then use the information gained to find ways ofsplicing it together without the discontinuities. You might think a simple cross-fade at the joinswould do the trick, and certainly it helps by eliminating clicks, but it is not enough. A musicalnote has a repeating waveform of fundamental frequency plus harmonics and if you splice thisat arbitrary points then the repeating waveform shape is upset with a jolt. In music editing youcan get away with a cross-fade splice here and there, but not 30 per second. What we wouldlike to do is somehow adjust our splice points in accordance with the frequency of the note sothat we splice exactly on a whole number of cycles and avoid any lurch in the waveform shape.
The techniques used for this divide into two categories, 'time domain' and'frequency domain'. The samples of a digital audio signal are considered to be'time domain' because the samples represent different points in time. To work inthe time domain is to work directly with these. In the time domain it is easy to identify the timeat which things happen but hard to identify frequency information. We can take a segment of sound and perform a discrete fourier transform (DFT) and this gives us a description of thatsegment as an array of data points where each point represents a different frequency : to workwith this data is to work in the frequency domain. In the frequency domain it is hard to identifythe time at which things happen but easy to identify frequency information.
'Time Domain' Techniques
The idea here is to identify the frequency of the note being played (there are varioustechniques for this) and splice only on whole numbers of cycles. This can work very well,especially if combined with transient-detection to avoid duplicating segments that have atransient in them. But there is a catch : it doesn't work when there are many notes being playedat the same time, it only works on one-note-at-a-time material. This makes it excellent for workingwith single note instruments or solo voice and it is used for the purpose in recording studios, butuseless for general purpose music which is polyphonic - many notes at a time. Transcribe! of courseneeds to work with polyphonic material so does not use this technique.
'Frequency Domain' Techniques
The problem with time domain techniques on polyphonic material is that if we choose a splicepoint to avoid discontinuity on one of the notes present then this splice point is unlikely to be suitablefor the other notes present. What we really want is to separate out the various notes and handleeach one differently. 'Phase alignment' is what we are talking about here. Thephase of a repeating waveform means, exactly what point in its repetition cycle has it reached?If we splice and the waveform has the same phase on both sides of the splice point then we haveno discontinuity in the waveform's repeating shape. But if the phase is different on either side,the waveform shape will lurch and not sound good.
The DFT (discrete fourier transform) tells us the amplitude and phase of each of the frequencycomponents present in the segment we DFT, and the fun part is that while we have the signal inthe frequency domain we can adjust the phases of the various frequencies independently so asto make them right for the splice point we are using. Then we use the inverse discrete fouriertransform (IDFT) to convert this back into the time domain, and use the resulting segment for thesplice. This is the 'Phase Vocoder', so called because there was once a weird studioeffect unit called the Vocoder which split a signal into maybe 8 frequency bands using analoguefilters, then applied envelope information from another source to modulate these bands. The phasevocoder is a bit like that except it preserves phase information too, hence the name. /garageband-08-free-download.html.
Transcribe! has always used a phase vocoder as do most programs which slow downpolyphonic music in real time. The FFT (fast fourier transform) algorithm makes it possible tocompute DFTs fast enough for this. However the phase vocoder has its difficulties too.
Perhaps the biggest problem with the phase vocoder is the question, how large shouldthe analysis segments be? (the segments we take from the input signal and apply the DFT to).To get accurate analysis of frequencies it is necessary for the segment to contain several(at least 3 or 4) full cycles of the lowest note we might see. If we expect notes down to say30Hz (not unreasonable) then this means segments of a tenth of a second. Unfortunately thisis a far larger segment than we would like to use at higher frequencies and results in severesmearing of transients at large slowdown ratios.
Djay 2 Slow Down Without Affecting Pitch Song
The answer to this is to use 'multiresolution analysis' where we split the signalinto several frequency channels and use a different segment size for each channel. Howeverthe phase computations are already quite tricky even for a single channel and if we havemultiple channels then we must also synchronise the phase between the channels or horriblethings happen in the crossover areas where one channel takes over from another. Prior toversion 5.2 Transcribe! offered two slowdown techniques - 'whole numbers'which used a two-channel multiresolution approach but only allowed whole number slowdownratios because this makes phase synchronisation between channels much easier, and'continuous' which allowed continuously variable speed but used only a singlechannel.
From version 5.2, Transcribe!'s slowdown incorporates the best features of both previous techniques, and more besides.It uses 5 channels running at consecutively lower sampling rates and using larger analysissegment sizes as the frequencies get lower. It allows continuously variable slowdown whilesynchronising phase between adjacent channels in the crossover zones.
Transcribe! version 7.2 has further improvements for the handling ofpercussive sounds and also gives a steadier sound thanks to improved handlingof phase. I think it soundspretty good and hope that you agree. If you haven't already tried Transcribe! then youcan download it for a 30 day free trial, and hear for yourself.
If You Want to Know More
Visit Google and search for'phase vocoder'.
© 2001-2018 Andy Robinson, Seventh String Software
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