Transforming Audio Files

The audio transformer acts as a dynamic editor to permit you to change the properties of an entire audio file or a selected range of samples within the file. Select from the operations on one tab per session. To use tools from multiple tabs, edit the original then re-edit the target. Visit the samples page to hear examples of transformed files.

Amplitude Settings

• Invert switches the channels left to right and right to left.
• Reverse sets the file to play backwards.
• Silence reduces the audio to zero within the selected range of samples.
• Amplify will increase or decrease the volume of the loaded file or within the selected range of samples.
• Normalize to raise the file's volume so that the highest level sample in the file reaches the defined level.
• Fade in (soft to loud) or fade out (loud to soft). The audio data will be linearly faded from starting amplitude Left to ending amplitude Right.
• Vibrato creates a cyclical changing of the defined frequency of the input signal to the defined depth of the vibrato effect. Depth varies from 0 (no effect) to 100 (maximum effect). 
• Compressor/Expander - A compressor is a dynamic volume regulator. Gain will be reduced when the signal amplitude is high thus making louder passages softer and reducing the dynamic range. An expander increases the dynamic range of a signal so that low level signals are attenuated while the louder portions are neither attenuated nor amplified. Time is calculated based upon root-mean-square calculation, in ms. Typically RMS Time is equal to 100 ms.

Delay Settings

• Delay/Phaser - Delay is an echo effect that replays what you have played ones or several times after a certain period of time. It's something like the echoes you might hear shouting against a wall. The  phaser achieves its distinctive sound by creating one or more notches in the frequency domain that eliminate sounds at the notch frequencies.

Flanger/Chorus - Flanging is created by mixing a signal with a slightly delayed copy of itself, where the length of the delay is constantly changing. It is actually one specific type of phasing.

Flanging is a special case of the chorus effect: it is created in the same way that chorus is created. Typically, the delay of the echo for a flanger is varied between 0ms and 5ms at a rate of 0.5Hz. In days gone by, flanging was created by sound engineers who put their finger onto the tape reel's flange, thus slowing it down. Two identical recordings are played back simultaneously, and one is slowed down to produce a "whooshing" sound, like the sound is pulsating.

The chorus effect is so named because it makes the recording of a vocal track sound like it was sung by two or more people singing in chorus. This is achieved by adding a single delayed signal (echo) to the original input.

The Chorus differs from the Flanger in the amount of delay that is used. The delay times in a Chorus are larger than in a Flanger, usually somewhere between 20 ms. and 30 ms. (the Flanger's delay usually ranges from 1 ms. to 10 ms.) This longer delay doesn't produce the characteristic sweeping sound of the Flanger. The Chorus also differs from the Flanger in that there is generally no feedback used.

• Reverb is used to simulate the acoustical effect of rooms and enclosed buildings. The sound heard at any given time is the sum of the sound from the source, as well as the reflected sound. An impulse (such a hand clap) will decay exponentially.

Filters

Audio frequencies can be filtered out (attenuated) or conversely permitted by the settings within the filter. In order to apply a filter you must select a range of samples. This can be all samples in the file or only a small piece. Note that filters invoked from the main menu use presets while the buttons on the toolbar permit abnormal parameter adjustment.

Descriptions of the standard filters are listed below.

• Low Pass filter allows all frequencies lower than the cutoff frequency to pass through unaffected.
• High Pass filter allows all frequencies higher than the cutoff frequency to pass through unaffected.
• The Bandpass filter consists of a Low Pass and a High Pass filter combined to allow only frequencies falling within a certain range, usually mid-range. Set the pass frequency with the left slider as a reciprocal of the loaded file's frequency. i.e.: frequency/slider value. Set the steepness of the frequency curve with the right slider in a range of 50 to 100.

When the bandpass filter is invoked from the main menu the frequency is preset at the frequency of the source file divided by 200 with a feedback gain of 100.

When the bandpass filter button on the toolbar is used, the interface is presented allowing you to adjust the bandpass frequency and feedback gain (pitch steepness).

Use the Bandpass filter for recordings such as conversions in noisy rooms to eliminate both high and low frequency sound above and below conversational tones.

• Notch filter (Band Reject) is the opposite of the band pass filter which attenuates frequencies around the cutoff point and passes the rest. Sometimes the two are applied in series to cut or boost at the selected frequency.
• Low Shelf filter cuts or boosts the frequencies below the cutoff point. Shelf filters are used to change a broad spectrum of sound rather than attenuating unwanted frequencies.
• High Shelf filter cuts or boosts the frequencies above the cutoff point. Shelf filters are used to change a broad spectrum of sound rather than attenuating unwanted frequencies.
• The Fast Fourier Transition filter smoothes amplitude transition at various frequencies. Points set above zero amplify and filter, points set below zero attenuate and filter.

Graphics equalizer

The graphics equalizer is like a bass and treble control for various frequency ranges. With it you can remove unwanted sounds such as hum or enhance the selection.

Effects

• Resample to recalculate the samples in your sound file at a different rate than the file was originally recorded. If a file is resampled at a lower rate, sample values are removed from the sound file, decreasing its size, but also decreasing its available frequency range and possibly introducing aliasing. Resampling to a higher sample rate, often interpolates extra sample values into the sound file. This increases the size of the sound file and may or may not increase the quality.
• Stretch/Shrink increases or decreases the playing time of the file.