This software was built using the awC++ SDK

Any Time v.1.1

Time - Pitch - Sample rate converter

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Runs under:  Windows 10 / 8 / 7 / Vista / XP.

Step 1 — Input files

Step 2 — Processing options

Step 3 — Output options

Step 4 — Calculate output

Any Time  lets you independently take control of the duration, pitch, and sample rate of an audio recording.

Also see the Audio quality page for a few more in-depth descriptions…

What is not not?

Awards & Reviews…

Examples of how you can use it:


Screenshot 1
Any Time - Step 1: Select input files

Screenshot 2
Any Time - Step 2: Select processing options

Screenshot 3
Any Time - Step 3: Select output options

Screenshot 4
Any Time - Step 4: Calculate output data

Sample rate:

There are many algorithms for performing sample rate conversion of an audio recording (a.k.a. resampling). These vary widely in complexity and in audio quality. The simplest (e.g. 'polynomial interpolation algorithms' commonly used in synthesizers) are CPU-efficient but do not perform any filtering of the sound before downsampling (converting from a higher to a lower sample rate). Frequencies left from the original recording above the frequency-limit of the downsampled recording will result in very undesirable 'aliasing noise'. It is therefore very important to filter out those high frequencies before downsampling. The ideal filter would be a 'brick filter' that cuts off everything above the new frequency-limit and leaves everything below it intact. Unfortunately such a filter is not practical to design — every real world filter will have a 'slope' at the cut-off frequency. While you don't want to leave any information above the new frequency limit, you don't want to remove any more than necessary of the frequencies below the limit either. This is one of the two challenges of sample rate conversion. The other is minimizing distortion. For both problems, in general you can say that the more CPU power you throw at the problem, the better audio quality you can get. Let's have a look at a real world example:

Downsampling example

Here white noise (noise with a flat spectrum) generated at 48000 Hz has first been downsampled to 32000 Hz (which means a frequency limit of 16 KHz) and then upsampled to 48000 Hz again using various algorithms. All calculations have been done using 32-bit floating in order to avoid quantization issues.

High frequency range extrapolation:

In many applications a low sample rate is used as a necessity for reducing file size or transmission band-width requirements (e.g. normal phone-switches use 8000 Hz). The idea of trying to 'recreate' lost higher frequency components is not new, but it is not an easy problem. Of course, information theory says that this is impossible in the 'generic' case where nothing is assumed about the properties of the signal. But it is certainly possible to use different heuristic and statistical methods to try to come up with an algorithm for high frequency extrapolation (a.k.a. synthetic bandwidth extension) that subjectively improves the sound even though it may not provide a perfect 'recreation' of the original. A well know example is the 'Spectral Band Replication' algorithm (SBR) used in 'mp3pro' and 'AAC+'. There does not seem to be much released information about how the SBR algorithm actually works, but from what public information we have been able to obtain we guess that it works by matching the low-frequency spectrum against a database of spectrums collected from many types of material. The decoder then selects and adds the high-frequency components from one entry in the database. The file would then add 'hints' that do not take up much space, but which helps the decoder make a better selection from the database. One potential problem with this approach is that it may not ensure a harmonic relationship between the added high frequency components and the 'real' lower frequency spectrum (musical tones tend to have harmonic overtones — that's what makes them sound good to our ears). The proprietary algorithm that we have developed for Any Time use an entirely different approach. Very broadly speaking it, analyzes the existing frequency spectrum and tries to identify the 'fundamental frequencies' of the sound sources (much like our instrument tuner software), it then adds harmonic series of overtones. The real tricky issue ishow to identify these frequencies, how to determine the proper amplitudes for the overtones, how to handle non-harmonic contents, and how to do this in a manner that is stable over time.

HF extrapolation example

The graph shows a 44100 Hz synth pad that has first been downsampled to 8000 Hz (i.e. only frequencies below 4000 Hz have been retained from the original) and then 'restored' to 44100 Hz using Any Time's high frequency range extrapolation option.

Pitch scaling and formant correction:

Pitch scaling moves the 'pitch' of a recording up or down without changing the sample rate or the recording length. This is done by breaking down the sound into separate frequency components, scaling them by the desired factor, then re-synthesizing' the sound. Simply changing the playback speed has the same effect on the pitch - but that also changes the playback time by the same factor as the pitch. One side-effect both have in common is the so called "Mickey mouse" effect — i.e. speech (and music too) sound 'tinny' when pitched up, or 'boomy' when pitched down. This happens because most instruments — the human voice included — works by in one way or another exciting a 'resonance body' and this resonance body has a set of 'resonance frequencies' near which sound is better amplified. This is seen as 'hills', known as 'formants', if you look at a frequency-amplitude graph. When playing notes of different pitches on an instrument (or whatever is producing the sound), the resonance frequencies will remain fixed. But when pitch-scaling, you will also change the resonance frequencies by the same factor as you scale all the frequency components — this is why the 'character' of the instrument changes (e.g. from a normal voice character to a Mickey-mouse character voice). The solution is to apply what is commonly called 'formant correction'. This entails somehow analyzing the 'frequency envelope' of the sound (the overall 'shape' of the frequency graph — a good example BTW of something that is much, much easier for a human to do than for a computer) and then re-enforcing it on the pitch-scaled sound. The result is very often a sound with a much more 'natural' character!

Pitch scaling example

The graph shows a short segment of a violin recording that has been pitch scaled by a factor 1.2x.


Time stretching changes the length of a recording without changing the sample rate or the pitch of the sound. This is equivalent to doing both pitch scaling and resampling at the same time — so the same comments apply as for those two operations. A mathematical necessity when time-stretching, or when pitch scaling, is that different frequencies get different amounts of 'phase shift' when they are scaled. When time-stretching by non-integer factors this may sometimes be especially noticeable because an you can get a pronounced 'wah-wah' effect (amplitude modulation due to cancellation). Any Time employs two methods to try to deal with this, one is a feature to 're-synchronize' the phase at larger amplitude increases. The other is an optional feature to 'enforce the original volume envelope'. This will analyze the overall 'volume envelope' of the original recording and 'force' this back onto the processed audio — much like the formant correction but in the time domain instead of in the frequency domain. The side effect of that is that it will not preserve the spectral characteristics of the sound since 'soft frequencies components' are lifted up to compensate whenever 'loud frequency components' cancel each other out. If you prefer this, or not, is a personal choice — but first try without this option and if you are bothered by a 'wah-wah', then try turning it on.

Audio mastering:

In the high quality audio studios of today, recording, mixing and effects processing are often done at a high sample rate and bit depth (e.g. 96000 Hz, 32-bit floating point). When 'mastering it', i.e. preparing it for the final lower resolution distribution medium (e.g. 44100 Hz, 16-bit audio CD format), it is important to get everything right in order to maintain maximum audio quality.

List of file formats supported by Any Time…

Click on one of the links below to start downloading:



Limitations of the Trial version

Audio codec add-ons…

Any Time is commercial software marketed as Shareware.

This means that you get to "try it before you buy it". If you find that you like it and wish to continue using it past the 30 day free trial period, then you need to buy a license.

There's also more some incentives for buying it:

Buy it on-line here:

Payments are handled by PayPal.
Most credit cards are accepted.
EU-customers:  VAT will be added the price.

License and delivery terms:

After we have received your order, you will be sent an email containing a registration code. This is your license key that unlocks the trial version into the full version. Please note:   The code is normally sent within 48 hours, but not immediately (also, do check your "spam" or "junk" folders if you don't find it in your in-box).

What you buy is a "single user" license. It is valid for the current program version, and for any minor version number updates (major version updates may require an upgrade fee). You are allowed to install it on more than computer, but you are not allowed to "borrow" it to other persons. The license is personal and can not be transferred or resold.

Thank you for your order!

If everything went fine with the PayPal transaction, an email containing your reg-code and further instructions should arrive within the next 48 hours. Please be patient, orders are manually verified before delivery. If you don't see an email, be sure to check you junk-mail folder before contacting support.

Revision history for Any Time…

Awards & Reviews

ShareUp: 5 stars
SofoTex: 5 stars
Download3000: 5 star rating
Softsia: 5 stars
GeekFiles: 5/5 stars 'Exceptional Product'
MaxxDownload: 5 stars
Soft3k 5 stars
NewFreeDownloads: 5 stars
Ivertech: 5 stars
Download2Easy 5 stars + Editor's choice
BlueSofts 5 stars + Editor's choice
More to come… :-)