The fundamental wave is the one that gives a string its pitch. But the string is making all those other possible vibrations, too, all at the same time, so that the actual vibration of the string is pretty complex. The other vibrations (the ones that basically divide the string into halves, thirds and so on) produce a whole series of harmonics. 8-voice diatonic pitch shifter based on technology from Eventide’s flagship H8000 effects processor with panning, delay, detune, and feedback to create beautiful harmonies and sequences. Randomize function randomizes an amount of micro pitch shift resulting in simulation of natural character for each of the 8 voices.

1. Stringed 2 8 – Shift Pitch And Manipulate Tempos Differences
2. Stringed 2 8 – Shift Pitch And Manipulate Tempos De
3. Stringed 2 8 – Shift Pitch And Manipulate Tempos Em

Time & Pitch

Time & Pitch uses iZotope’s sophisticated Radius™ algorithm to give you independent control over the length and pitch of your audio. It is useful for retuning audio to fit in a mix better, or adjusting the length of audio to deal with BPM or time code changes.

Time & Pitch’s Pitch Contour tab can be used for faster pitch shifting with the ability to correct variations in pitch over time.

iZotope Radius

iZotope Radius™ is a world-class time-stretching and pitch-shifting algorithm. You can easily change the pitch of a single instrument, voice, or entire ensemble while preserving the timing and acoustic space of the original recording. iZotope Radius is designed to match the natural timbres even with extreme pitch shifts.

Algorithm

You should use Solo mode only when processing a single instrument with a clearly defined pitch. The human voice is a good candidate for solo mode, as are most stringed instruments, brass instruments, and woodwinds. For most other types of source material, Radius mode will usually offer better results. If speed is important, use the Radius RT mode.

Solo

In Solo mode, the adaptive window size can significantly affect the quality of Radius's output. If the adaptive window size is too small, you will hear a squeaking noise which sounds like the pitch of the audio is changing very rapidly. If the adaptive window size is too large then the sound will become grainy as you will begin to hear portions of it being repeated.

A good approach is to start with the default window size of 37 ms. If the results are unsatisfactory, increase the window size until the squeaking noise described above does not occur. If you cannot get the distortion to disappear, switch to Radius mode for processing.

Lower pitched instruments and voices may require a longer adaptive window size than the default, but very long adaptive window sizes can cause audible repeating slices of audio.

Formant Correction

Formants are the resonant frequency components of voice that tend to be perceived as characteristics like age and gender. You can shift formants independently of pitch and time by enabling Shift Formants.

Stringed 2 8 – Shift Pitch And Manipulate Tempos Differences

Typically you will leave the Formant Shift Strength set to 1 (full strength) and the Formant Shift Semitones set to 0. If you hear what sounds like an EQ adjustment to your audio, you can try lowering the strength to reduce this artifact. To achieve special effects, for example to change the perceived gender of a human voice, try adjusting the semitones to a value other than 0.

Stretch & Shift Controls

Stretch Ratio

Determines how much the resulting audio will be stretched in time. Values between 12.5% and 100% will cause the audio to speed up without affecting pitch, resulting in a shorter audio file. Values between 100% and 800% will cause the audio to slow down without affecting pitch, giving you a longer audio file.

BPM Calculator

If you are using Radius to process audio for a tempo change, you can also adjust the stretch ratio with the BPM Calculator.

Pitch Shift

Controls the amount of pitch shifting up or down that will be applied to the audio.

Algorithm

The Algorithm drop-down menu has three options:

• Radius — designed to work well with polyphonic material such as mixes with more than one instrument, as well as non-harmonic material such as drum loops or rhythmic audio. This is the highest-quality option for most sources.
• Solo Instrument — designed for monophonic pitched material such as a stringed instrument or human voice.
• Radius RT — good quality, polyphonic, but faster than Radius.

Transient Sensitivity

Determines the algorithm’s handling of transient material. Higher values will result in better preservation of individual transients after processing.

When stretching percussive material, you usually want transient sensitivity set to its default value of 1. If transients in your audio are being 'smeared', a higher value of 2 will tighten up transience at the expense of incurring heavier processing on non-transient audio.

Bowed instruments such as the violin and cello are especially affected by the transient sensitivity setting. If you hear a stuttering artifact, lower the transient sensitivity to eliminate it.

Noise Generation (Radius mode only)

Helps noisy material (like sibilance or snare drums) sound more natural when processed.

This control will generate noise instead of stretching the noise that is already present in the signal and creating new tones. Higher values of the noise generation parameter will cause Radius to generate noise more often, but can cause some phase artifacts.

Pitch Coherence (Radius mode only)

Controls the preservation of the natural timbre of the processed audio.

The Pitch coherence control in the Radius control panel helps preserve the timbre for pitched solo voices, such as human speech, saxophone or vocals. While traditional vocoders can smear these signals in time and randomize phase, the pitch coherence parameter of Radius preserves phase coherence for these signals.

High values of pitch coherence will avoid phasiness in Radius's output at the expense of roughness (modulation) in processed polyphonic recordings. Try turning this up for better results if you’re processing a solo voice or a small group of related instruments.

Phase Coherence (Mix mode only)

Preserves the coherence of phase between the left and right channels of the processed audio.

This should be increased if there's any change in the perceived stereo image after using Radius. It can be decreased when processing a multichannel signal where different channels contain completely different instruments.

Adaptive Window Size (ms) (Solo mode only)

Adjusts the window size in milliseconds of Radius' Solo algorithm.

If the adaptive window size is too small, you will hear a squeaking noise which sounds like the pitch of the audio is changing very rapidly. If the adaptive window size is too large then the sound will become grainy as you will begin to hear portions of it being repeated.

Increase this if you have trouble getting good results pitching or stretching low-pitched instruments or voices.

Shift Formants

Processes formant frequencies independently of other pitch and time processing.

When this option is enabled, formant frequencies can be shifted independently of other pitch shifting performed by Radius.

When Radius performs pitch-shifting without Formant Correction, it will shift these resonant frequencies along with the rest of the audio.

• Strength — adjusts the amplitude strength of the formant correction filter.
• Shift — how much formant frequencies are shifted. Typically this control can be set to 0, which leaves the formant frequencies unshifted during processing. Adjust this control to fine-tune the formant correction algorithm or for special effects.
• Width — controls the bandwidth of the formant detection filter. Smaller values of this control will offer more precise formant correction in the processed audio. Higher values will include a wider band of formant frequencies.

Pitch shifting single instruments (especially bass instruments) can benefit from some adjustments to formant correction. Try enabling formant correction and moving the strength between 0.1 and 0.2. Move the Formant Correction semitones part of the way towards your pitch shift amount. For example, if you're pitch shifting +4 semitones, move the Formant Correction Semitones between 2 and 3. This can help bring back subtle percussive elements in the original source material.

The formant frequencies of the human voice can actually shift slightly when we sing. You can use the Formant Correction Semitones control to compensate for this. For example, if pitch shifting a human voice by +7 semitones, try setting the Formant Correction semitones between 0 and +2 for more natural results.

Pitch Contour Controls

The Pitch Contour mode of the Time & Pitch module lets you change the pitch of a selection over time. This can be used to quickly correct small pitch variations or gradual pitch drifts over time.

The Pitch Contour changes pitch by continuously changing the playback speed of the audio. The effect is similar to speeding up or slowing down a record or tape deck while it is playing back.

Because the Pitch Contour uses resampling to synchronously change time and pitch, it cannot be used to adjust pitch without also adjusting time.

Pitch Contour

The horizontal axis shows the length of your current selection. If you have no selection, the horizontal axis represents the entire length of your file.

The vertical axis shows the amount of pitch shifting that will be applied. A curve through the top half of the display will create a higher shift in pitch and shorten the audio correspondingly. A curve through the lower half of the display will create a lower shift in pitch and lengthen the audio correspondingly.

You can correct a gradual pitch drift over time by adjusting the points at the far left or right of the display, drawing a straight sloping line from the beginning of your selection to the end. These points are locked to the vertical axis.

Clicking on the contour display will create a new pitch node. You can create up to 20 pitch nodes to achieve very complicated pitch shifts.

Clicking and dragging a pitch node to move it around will change the pitch curve.

Double clicking on a pitch node will set its value to 0 (no change at that point).

Right clicking on a pitch node will delete it.

Holding control/command while dragging will give you fine control over a pitch node’s position.

Smoothing

Larger values create a smoother pitch curve when multiple pitch nodes are present. This is useful when correcting a nonlinear change in pitch.

Reset

Clears all pitch nodes and returns the Smoothing control to its default value.

Time stretching is the process of changing the speed or duration of an audio signal without affecting its pitch. Pitch scaling is the opposite: the process of changing the pitch without affecting the speed. Pitch shift is pitch scaling implemented in an effects unit and intended for live performance. Pitch control is a simpler process which affects pitch and speed simultaneously by slowing down or speeding up a recording.

These processes are often used to match the pitches and tempos of two pre-recorded clips for mixing when the clips cannot be reperformed or resampled. Time stretching is often used to adjust radio commercials^[1] and the audio of television advertisements^[2] to fit exactly into the 30 or 60 seconds available. It can be used to conform longer material to a designated time slot, such as a 1-hour broadcast.

Resampling[edit]

The simplest way to change the duration or pitch of a digital audio clip is through sample rate conversion. This is a mathematical operation that effectively rebuilds a continuous waveform from its samples and then samples that waveform again at a different rate. When the new samples are played at the original sampling frequency, the audio clip sounds faster or slower. Unfortunately, the frequencies in the sample are always scaled at the same rate as the speed, transposing its perceived pitch up or down in the process. In other words, slowing down the recording lowers the pitch, speeding it up raises the pitch. This is analogous to speeding up or slowing down an analogue recording, like a phonograph record or tape, creating the Chipmunk effect. Using this method the two effects cannot be separated. A drum track containing no pitched instruments can be moderately sample-rate converted for tempo without adverse effects, but a pitched track cannot.

Frequency domain[edit]

Phase vocoder[edit]

One way of stretching the length of a signal without affecting the pitch is to build a phase vocoder after Flanagan, Golden, and Portnoff.

Basic steps:

1. compute the instantaneous frequency/amplitude relationship of the signal using the STFT, which is the discrete Fourier transform of a short, overlapping and smoothly windowed block of samples;
2. apply some processing to the Fourier transform magnitudes and phases (like resampling the FFT blocks); and
3. perform an inverse STFT by taking the inverse Fourier transform on each chunk and adding the resulting waveform chunks, also called overlap and add (OLA).^[3]

The phase vocoder handles sinusoid components well, but early implementations introduced considerable smearing on transient ('beat') waveforms at all non-integer compression/expansion rates, which renders the results phasey and diffuse. Recent improvements allow better quality results at all compression/expansion ratios but a residual smearing effect still remains.

The phase vocoder technique can also be used to perform pitch shifting, chorusing, timbre manipulation, harmonizing, and other unusual modifications, all of which can be changed as a function of time.

Sinusoidal spectral modeling[edit]

Another method for time stretching relies on a spectral model of the signal. In this method, peaks are identified in frames using the STFT of the signal, and sinusoidal 'tracks' are created by connecting peaks in adjacent frames. The tracks are then re-synthesized at a new time scale. This method can yield good results on both polyphonic and percussive material, especially when the signal is separated into sub-bands. However, this method is more computationally demanding than other methods.^{[citation needed]}

Time domain[edit]

SOLA[edit]

Rabiner and Schafer in 1978 put forth an alternate solution that works in the time domain: attempt to find the period (or equivalently the fundamental frequency) of a given section of the wave using some pitch detection algorithm (commonly the peak of the signal's autocorrelation, or sometimes cepstral processing), and crossfade one period into another.

This is called time-domain harmonic scaling^[5] or the synchronized overlap-add method (SOLA) and performs somewhat faster than the phase vocoder on slower machines but fails when the autocorrelation mis-estimates the period of a signal with complicated harmonics (such as orchestral pieces).

Adobe Audition (formerly Cool Edit Pro) seems to solve this by looking for the period closest to a center period that the user specifies, which should be an integer multiple of the tempo, and between 30 Hz and the lowest bass frequency.

This is much more limited in scope than the phase vocoder based processing, but can be made much less processor intensive, for real-time applications. It provides the most coherent results^{[citation needed]} for single-pitched sounds like voice or musically monophonic instrument recordings.

High-end commercial audio processing packages either combine the two techniques (for example by separating the signal into sinusoid and transient waveforms), or use other techniques based on the wavelet transform, or artificial neural network processing^{[citation needed]}, producing the highest-quality time stretching.

Frame-based approach[edit]

In order to preserve an audio signal's pitch when stretching or compressing its duration, many time-scale modification (TSM) procedures follow a frame-based approach.^[6]Given an original discrete-time audio signal, this strategy's first step is to split the signal into short analysis frames of fixed length.The analysis frames are spaced by a fixed number of samples, called the analysis hopsize${\displaystyle H_a\in \mathbb{N}}$.To achieve the actual time-scale modification, the analysis frames are then temporally relocatedto have a synthesis hopsize${\displaystyle H_s\in \mathbb{N}}$.This frame relocation results in a modification of the signal's duration by a stretching factor of${\displaystyle \alpha =H_s/H_a}$.However, simply superimposing the unmodified analysis frames typically results in undesired artifactssuch as phase discontinuities or amplitude fluctuations.To prevent these kinds of artifacts, the analysis frames are adapted to form synthesis frames, prior tothe reconstruction of the time-scale modified output signal.

Stringed 2 8 – Shift Pitch And Manipulate Tempos De

The strategy of how to derive the synthesis frames from the analysis frames is a key difference amongdifferent TSM procedures.

Speed hearing and speed talking[edit]

For the specific case of speech, time stretching can be performed using PSOLA.

While one might expect speeding up to reduce comprehension,Herb Friedman says that 'Experiments have shown that the brain works most efficiently if the information rate through the ears—via speech—is the 'average' reading rate, which is about 200–300 wpm (words per minute), yet the average rate of speech is in the neighborhood of 100–150 wpm.'^[7]

Speeding up audio is seen as the equivalent of speed reading.^[8][9]

Pitch scaling[edit]

These techniques can also be used to transpose an audio sample while holding speed or duration constant. This may be accomplished by time stretching and then resampling back to the original length. Alternatively, the frequency of the sinusoids in a sinusoidal model may be altered directly, and the signal reconstructed at the appropriate time scale.

Transposing can be called frequency scaling or pitch shifting, depending on perspective.

Stringed 2 8 – Shift Pitch And Manipulate Tempos Em

For example, one could move the pitch of every note up by a perfect fifth, keeping the tempo the same.One can view this transposition as 'pitch shifting', 'shifting' each note up 7 keys on a piano keyboard, or adding a fixed amount on the Mel scale, or adding a fixed amount in linear pitch space.One can view the same transposition as 'frequency scaling', 'scaling' (multiplying) the frequency of every note by 3/2.

Musical transposition preserves the ratios of the harmonic frequencies that determine the sound's timbre, unlike the frequency shift performed by amplitude modulation, which adds a fixed frequency offset to the frequency of every note. (In theory one could perform a literal pitch scaling in which the musical pitch space location is scaled [a higher note would be shifted at a greater interval in linear pitch space than a lower note], but that is highly unusual, and not musical^{[citation needed]}).

Time domain processing works much better here, as smearing is less noticeable, but scaling vocal samples distorts the formants into a sort of Alvin and the Chipmunks-like effect, which may be desirable or undesirable.A process that preserves the formants and character of a voice involves analyzing the signal with a channel vocoder or LPC vocoder plus any of several pitch detection algorithms and then resynthesizing it at a different fundamental frequency.

A detailed description of older analog recording techniques for pitch shifting can be found within the Alvin and the Chipmunks entry.

See also[edit]

others

• Dynamic tonality — the real-time changes of tuning and timbre for new chord progressions, musical temperament modulations, etc.

References[edit]

1. ^https://web.archive.org/web/20080527184101/http://www.tvtechnology.com/features/audio_notes/f_audionotes.shtml
2. ^http://www.atarimagazines.com/creative/v9n7/122_Variable_speech.php
3. ^Jont B. Allen (June 1977). 'Short Time Spectral Analysis, Synthesis, and Modification by Discrete Fourier Transform'. IEEE Transactions on Acoustics, Speech, and Signal Processing. ASSP-25 (3): 235–238.
4. ^McAulay, R. J.; Quatieri, T. F. (1988), 'Speech Processing Based on a Sinusoidal Model'(PDF), The Lincoln Laboratory Journal, 1 (2): 153–167, archived from the original(PDF) on 2012-05-21, retrieved 2014-09-07
5. ^David Malah (April 1979). 'Time-domain algorithms for harmonic bandwidth reduction and time scaling of speech signals'. IEEE Transactions on Acoustics, Speech, and Signal Processing. ASSP-27 (2): 121–133.
6. ^Jonathan Driedger and Meinard Müller (2016). 'A Review of Time-Scale Modification of Music Signals'. Applied Sciences. 6 (2): 57. doi:10.3390/app6020057.
7. ^Variable Speech, Creative Computing Vol. 9, No. 7 / July 1983 / p. 122
8. ^http://www.nevsblog.com/2006/06/23/listen-to-podcasts-in-half-the-time/
9. ^https://web.archive.org/web/20060902102443/http://cid.lib.byu.edu/?p=128

External links[edit]

• Time Stretching and Pitch Shifting Overview A comprehensive overview of current time and pitch modification techniques by Stephan Bernsee
• Stephan Bernsee's smbPitchShift C source code C source code for doing frequency domain pitch manipulation
• pitchshift.js from KievII A Javascript pitchshifter based on smbPitchShift code, from the open source KievII library
• The Phase Vocoder: A Tutorial - A good description of the phase vocoder
• How to build a pitch shifter Theory, equations, figures and performances of a real-time guitar pitch shifter running on a DSP chip
• ZTX Time Stretching Library Free and commercial versions of a popular 3rd party time stretching library for iOS, Linux, Windows and Mac OS X
• Elastique by zplane commercial cross-platform library, mainly used by DJ and DAW manufacturers
• Voice Synth from Qneo - specialized synthesizer for creative voice sculpting
• TSM toolbox Free MATLAB implementations of various Time-Scale Modification procedures
• Pitch Shifter Audio Tool Online pitch-shifting audio tool implemented by SoundTouch algorithm