Practice will never be boring again!ย
Phrase Trainer 2 – slowdowner, pitch shifter, looper and more
Introducing Phrase Trainer 2
From getting a great accuracy and speed to being able to play in all keys today’s musicians must master every element of their skills. Phrase Trainer 2 is here to help with every step on the way of progress. And thanks to the redesigned user interface, Phrase Trainer 2 lets you move seamlessly among the elements, instantly going from setting/adjusting loop points,isolating parts of the song, to speed and key training and back again. And do it all with a single smart tool.
Why struggle with the cluttered interfaces, poor navigation, and antiquated window management that plague most similar software? Phrase Trainer 2 interface gives you a clear and configurable creative space that adapts to your practice routine and is made for musicians by a musician.
“I have been using phrase trainer since its first version and I use it every day for myself and with my guitar students!”
-Andrew (USA)
Phrase Trainer 2 has been cleverly designed to make every aspect of practice even faster and more fluid. Turn on/off different tools as you need them, instantly set loops and switch between them, and navigate with unparalleled speed and precision with Timeline Zoom and big waveform display. Phrase Trainer 2 even remembers all settings for you automatically so you can quit fiddling with the software and get on with the business of practice or playing along just for the fun of it.
Boost your skills to the next level with Phrase Trainer 2.
Features
- Pitch shift in -12…+12 semitones
- Fine tuning in cents(1/100 of semitone)
- Slowdown/speedup from -50%(half) to 100% (double) speed
- Define as many loops ย per song as you want
- Zoom in and out of large waveform display
- Speed trainer, Pitch trainer
- Integrated file explorer, play list support, folder based playing, recent files and folders list
- Manipulate loops (Double loops,halve,shift…)
- One click saving,no boring “save” dialogs etc…
- MUSICIAN INTERFACE not USER INTERFACE !!!
-
Slow down music without changing pitch
- Play all standar audio files including midi files
- Export song as mp3,wav,ogg,wma
- Presets
- Keyboard shortcuts for all play-loop-pitch-tempo functions
Phrase Trainer 2 โ slowdowner, pitch shifter, looper and more
A phrase trainer is a tool used to help musicians learn and practice music. It is an effective way to train your ear and improve your playing skills. Phrase Trainer 2 is a modern version of the classic phrase trainer that allows you to slow down, speed up, loop, and adjust the pitch of music. This makes it easy to learn difficult passages and to practice them at your own pace.
The main features of Phrase Trainer 2 include a tempo control, a pitch control, and a looping feature. The tempo control allows you to slow down or speed up the music without affecting the pitch. This is great for learning difficult passages that need to be practiced slowly. The pitch control allows you to raise or lower the pitch of the music, making it easier to learn passages in different keys. The looping feature is perfect for practicing difficult passages over and over.
The phrase trainer is a valuable tool for any musician, from beginners to professionals. With Phrase Trainer 2, you can slow down, speed up, loop, and adjust the pitch of music to make your practice sessions more efficient and effective.
Aw, this was a really nice post. Spending some time and actual effort to produce a top notch article?
but what can I say? I procrastinate a whole lot and don’t manage to get nearly anything done.
It just crashes on my Windows 7.
Please provide a little more information so I can try to help
If I buy Phrase Trainer 2 can I install it on my studio PC and my laptop too that I usually take with me to practice with friends?
Sure thing ๐
Can you add shortcuts for navigation between regions? I own the software and like it by the way,thanks.
You can do that by using Tab and BackSpace to move forward and backward between regions and pressing Enter will select that region for looping.
I will,thanks ๐
Hey! Its nice that you have comments section now! I had the phrase trainer from the beginning and was very happy with it and this new version is a really big step,good work, keep it up!
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amplitude and phase of each of the
subject. It is a fairly general discussion but I also comment on the
gaps by duplicating bits of the
phases of the
the
the
2, Transcribe!’s slowdown incorporates the
Also transients are duplicated if they happen to be in a segment which gets used twice, creating a smeared-out effect which becomes very bad at high slowdown ratios
“Phase alignment” is what we are talking about here
sample rate – merely enables you to change the
pitch and speed together in a way that’s exactly analogous to varying the
It uses 5 channels running at consecutively lower sampling rates and using larger analysis segment sizes as the
segments we take from the
the
tonguing of a brass instrument, the
frequency domain
Are we aiming for a high quality realistic musical result, in which case it is reasonable to limit ourselves to small changes in speed (because large changes can never sound musically realistic anyway) or do we want large changes, in which case we must accept that it will sound “processed”?
the
playback head was circular and in fact had four playback heads equally spaced around the
Vocoder which split a signal into maybe 8 frequency bands using analogue filters, then applied envelope information from another source to modulate these bands
Suppose we have a sustained bass note at 31Hz (B, the
splice point we are using
particular methods which I chose to use in Transcribe!
crossover areas where one channel takes over from another
pitch, then resample to speed it back up to the
same time, it only works on one-note-at-a-time material
playback head which would not be the
tape was the
From version 5
various notes and handle each one differently
sound to some extent, then use the
bass note and the
If you haven’t already tried Transcribe! then you can download it for a 30 day free trial, and hear for yourself
frequency of the
speed and 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 than about 20 or 30% either way, as anything more than that stands practically no chance of sounding natural anyway. High quality programs in this area do indeed make the
I mostly discuss slowing down rather than speeding up here, but with a moment’s thought you can see that pretty much every issue discussed applies equally to both.
samples of a digital audio signal are considered to be “time domain” because the
There are also “modelling” techniques which attempt to analyse the
head rotated while the
This crude technique (whether implemented mechanically or digitally) is easy to do but has many problems with sound quality
slowdown method must generate more “cycles” of oscillation to fill up the
Most commercial speed or pitch change software is intended for music recording and editing applications – for instance changing the
input sound into short segments – typically in the
hammer of a piano, the
phase vocoder is a bit like that except it preserves phase information too, hence the
Pitch (technical discussion)
circumference
effort of locating transients and not modifying them, and many other sophisticated techniques. For this to work in real time you would usually be talking about a dedicated DSP processing effects unit.
We can take a segment of sound and perform a discrete fourier transform (DFT) and this gives us a description of that segment as an array of data points where each point represents a different frequency : to work with this data is to work in the
segment to contain several (at least 3 or 4) full cycles of the
the
2 Transcribe! offered two slowdown techniques – “whole numbers” which used a two-channel multiresolution approach but only allowed whole number slowdown ratios because this makes phase synchronisation between channels much easier, and “continuous” which allowed continuously variable speed but used only a single channel
information gained to find ways of splicing it together without the
signal into several frequency channels and use a different segment size for each channel
phase between the
same phase on both sides of the
phase of a repeating waveform means, exactly what point in its repetition cycle has it reached? If we splice and the
note being played (there are various techniques for this) and splice only on whole numbers of cycles
ability to slow down music (or speed it up) in real time without changing the
For instance if you want to raise the
frequencies get lower
speed without changing the
the
pitch. People sometimes ask how this is done so I have written this discussion of the
the
But there is a catch : it doesn’t work when there are many notes being played at the
the
the
Perhaps the
frequency domain it is hard to identify the
other hand we might want to hear the
Transcribe! version 7
overall speed was controlled by the
What we would like to do is somehow adjust our splice points in accordance with the
analysis segments be? (the
Transcribe!’s Slowdown
In music editing you can get away with a cross-fade splice here and there, but not 30 per second
Transcribe! has always used a phase vocoder as do most programs which slow down polyphonic music in real time
guitar strum could very easily both be happening at the
frequency components present in the
music down much more drastically – by a factor of up to 20 in fact – but “natural” sound is fortunately not such a priority, instead the
In the
frequency domain we can adjust the
sustained notes or silences between them. On the
the
sounds to be handled with different approaches given that the
high level description
the
attack, for instance if we analysing a player’s technique for teaching purposes, in which case we want to stretch everything equally.
“Time Domain” Techniques
phase vocoder is the
phase is different on either side, the
extra time. On the
relative speed of the
You can see how this involves playing certain little slices of tape twice as one head takes over from another
“Phase Vocoder”, so called because there was once a weird studio effect unit called the
DFT to)
If you have never tried it then you might think that once you have some music on your hard disk in digital form, it would be easy to change the
original speed while raising the
splice point then we have no discontinuity in the
tape moved past it and a brush contact underneath ensured that the
samples represent different points in time
answer to this is to use “multiresolution analysis” where we split the
resulting segment for the
one whose output was fed to the
the
trick, and certainly it helps by eliminating clicks, but it is not enough
biggest problem with the
desired handling is radically different. How can any slowdown method know which approach to use, or even separate out the
bass note vs. guitar strum example above is a tricky one but multi-resolution processing (see below) helps a lot.
pitch too
problem with time domain techniques on polyphonic material is that if we choose a splice point to avoid discontinuity on one of the
waveform shape
priority is to be able to hear clearly what’s happening. For this reason I regard it as more sensible – and easier – for Transcribe! to stretch everything equally.
Background
Prior to version 5
author) is intended to help musicians to transcribe music from recordings. It has the
other hand suppose we have a guitar strum where the
the
pitch – just a bit of resampling or something like that. But in fact it’s difficult. Resampling – changing the
techniques we can use to try to reduce these bad effects
other notes present
tape while the
hit of a drum. If we want the
various frequencies independently so as to make them right for the
splice
gaps
In the
note so that we splice exactly on a whole number of cycles and avoid any lurch in the
time domain it is easy to identify the
material at a high level and then reconstruct at slower speed from the
same if the
Transcribe! of course needs to work with polyphonic material so does not use this technique
segment we DFT, and the
the
handling of percussive sounds and also gives a steadier sound thanks to improved handling of phase
This can work very well, especially if combined with transient-detection to avoid duplicating segments that have a transient in them
playback amplifier
inverse discrete fourier transform (IDFT) to convert this back into the
To get accurate analysis of frequencies it is necessary for the
FFT (fast fourier transform) algorithm makes it possible to compute DFTs fast enough for this
beginning of each note, which is quite different from the
sound and as there are perhaps 30 splice points per second, this causes a dreadful warbling noise
rest of this discussion will be about some of the
Before we can decide what we should be aiming for we have to ask what our slowdown program will be used for. Here are some questions we must consider:
pitch which I won’t be discussing here
basic technique used by most slowdown methods whether “time domain” or “frequency domain” (see below) is to slice the
Transcribe! is intended as an aid for transcribing and needs to slow the
It allows continuously variable slowdown while synchronising phase between adjacent channels in the
head was rotating
signal in the
To work in the
best features of both previous techniques, and more besides
note to remain at 31Hz (31 cycles per second) so the
the
2 has further improvements for the
sound of the
problem of changing speed without changing pitch you can easily change pitch without changing speed by applying a touch of resampling afterwards
This makes it excellent for working with single note instruments or solo voice and it is used for the
pitch was controlled by the
But if the
speed without changing the
A musical note has a repeating waveform of fundamental frequency plus harmonics and if you splice this at arbitrary points then the
lowest note we might see
What Do We Really Want?
Apparently back in the
speed of the
crossover zones
DFT (discrete fourier transform) tells us the
pitch then you first lower the
input signal and apply the
I think it sounds pretty good and hope that you agree
Are we running in real time on a desktop computer in which case we must use efficient processing techniques, or are we running on expensive dedicated hardware or non-realtime, in which case we can use more sophisticated (slower) methods?
head which was currently in contact with the
However the
the
What we really want is to separate out the
discontinuities
frequency of the
repeating waveform shape is upset with a jolt
notes present then this splice point is unlikely to be suitable for the
waveform shape will lurch and not sound good
idea here is to identify the
range from a 100th to a 10th of a second – to spread those segments further apart in time, and to fill the
“Frequency Domain” Techniques
waveform has the
strum has about 30 note-attacks per second. If we slow this down we want the
pitch of a singer’s voice to correct an out-of-tune note, or adjusting the
joins would do the
Are we working on polyphonic material (many notes at a time) or just one note at a time? Different techniques may be preferred, depending.
You might think a simple cross-fade at the
time domain, and use the
segments either side – a sort of “copy and paste” into the
slowed-down music to sound like a real player on a real instrument who simply happens to be playing more slowly then we must not stretch these “transients” but only stretch the
details of the
If we expect notes down to say 30Hz (not unreasonable) then this means segments of a tenth of a second
Slow Down Music Without Changing the
note-attacks to be spread further apart while there are still 6 of them. So here we have two examples of sonic events which happen at about 30 per second, for which the
Basic Technique
time at which things happen but hard to identify frequency information
time at which things happen but easy to identify frequency information
This is the
However the
There are also more direct ways of changing the
way, once you solve the
phase vocoder has its difficulties too
fun part is that while we have the
waveform’s repeating shape
These can be good on certain material but I won’t be discussing them here
the
sustained note – for instance the
guitarist smoothly strums all 6 strings in about a fifth of a second – a moderate speed strum. In this case the
speed of an analogue tape recorder or vinyl record player. Halve the
channels or horrible things happen in the
steam age you could get tape recorders which implemented this technique mechanically
Then we use the
basic idea is, we must analyse the
tape past the
By the
same time?
purpose in recording studios, but useless for general purpose music which is polyphonic – many notes at a time
splice points introduce discontinuities in the
techniques used for this divide into two categories, “time domain” and “frequency domain”
pitch goes down an octave.
Unfortunately this is a far larger segment than we would like to use at higher frequencies and results in severe smearing of transients at large slowdown ratios
phase computations are already quite tricky even for a single channel and if we have multiple channels then we must also synchronise the
Most instruments have a clearly defined start-up noise at the
question, how large should the
open bottom string of a 5 string bass). When we slow this down we would clearly want the
time domain is to work directly with these
name
program “Transcribe!” (of which I am the
the