Information about Audio Timescale Pitch Modification
Time stretching is the process of changing the speed or duration of an audio signal without affecting its pitch.
Pitch scaling or pitch shifting is the reverse: the process of changing the pitch without affecting the speed. There are also more advanced methods used to change speed, pitch, or both at once, as a function of time.
These processes are used, for instance, to match the pitches and tempos of two pre-recorded clips for mixing when the clips cannot be reperformed or resampled. (A drum track could be moderately resampled for tempo without adverse effects, but a pitched track could not). They are also used to create effects such as increasing the range of an instrument (like pitch shifting a guitar down an octave).
Basic steps:
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.
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 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, producing the highest-quality time stretching.
Transposing can be called pitch scaling or pitch shifting, depending on perspective.
For example, one could move the frequency 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 "pitch scaling", "scaling" (multiplying) the frequency of every note by 3/2.
Musical transposition preserve 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).
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.
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These processes are used, for instance, to match the pitches and tempos of two pre-recorded clips for mixing when the clips cannot be reperformed or resampled. (A drum track could be moderately resampled for tempo without adverse effects, but a pitched track could not). They are also used to create effects such as increasing the range of an instrument (like pitch shifting a guitar down an octave).
Resampling
The simplest way to change the duration or pitch of a digital audio clip is to resample it. 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. In other words, slowing down the recording lowers the pitch, speeding it up raises the pitch, and the two effects cannot be separated. This is analogous to speeding up or slowing down an analog recording, like a phonograph record or tape, creating the chipmunk effect.Phase vocoder
Basic steps:
- 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;
- apply some processing to the Fourier transform magnitudes and phases (like resampling the FFT blocks); and
- perform an inverse STFT by taking the inverse Fourier transform on each chunk and adding the resulting waveform chunks.
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.
Time domain
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 or the synchronized overlap-add method 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. For a 120 bpm tune, use 48 Hz because 48 Hz = 2,880 cycles/minute = 24 cycles/beat * 120 bpm.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 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, producing the highest-quality time stretching.
Pitch scaling
These techniques can also be used to transpose an audio sample while holding speed or duration constant.Transposing can be called pitch scaling or pitch shifting, depending on perspective.
For example, one could move the frequency 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 "pitch scaling", "scaling" (multiplying) the frequency of every note by 3/2.
Musical transposition preserve 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).
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.
See also
External links
- 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
- The Phase Vocoder: A Tutorial - A good description of the phase vocoder
- New Phase-Vocoder Techniques for Pitch-Shifting, Harmonizing and Other Exotic Effects
- A new Approach to Transient Processing in the Phase Vocoder
- PSOLA Synthesis, SOLAFS Synthesis - Two specific methods of time domain TDHS or SOLA processing.
- Audio Engineering Society
- Original E2 article (http://everything2.com/index.pl?node_id=1074923)
- DSPdimension: DIRAC library
- zplane.development: élastique SDKs
- http://www.bdti.com/faq/dsp_faq.htm - comp.dsp FAQ
- SoundTouch library - An open-source implementation of time/pitch scaling algorithms. SoundStretch came from here. Used in Audacity.
- PICOLA and TDHS
- wavMasher - Time and pitch scaling software
- PaulStretch A program that works for extreme time stretching (like 50x), only
- 4 Band Shifter An open source VST plugin based on Bernsee's code that shifts the pitch on 4 independent, user-definable frequency bands.
Audio signal processing, sometimes referred to as audio processing, is the processing of a representation of auditory signals, or sound. The representation can be digital or analog.
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Pitch is the perceived fundamental frequency of a sound. While the actual fundamental frequency can be precisely determined through physical measurement, it may differ from the perceived pitch because of overtones, or partials, in the sound.
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The term digital signal is used to refer to more than one concept. It can refer to discrete-time signals that are digitized, or to the waveform signals in a digital system.
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For resampling methods in statistics, see .
Resampling is the digital process of changing the sample rate or dimensions of digital imagery or audio by temporally or areally analysing and sampling the original data...... Click the link for more information.
An analog or analogue signal is any time continuous signal where some time varying feature of the signal is a representation of some other time varying quantity. It differs from a digital signal in that small fluctuations in the signal are meaningful.
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gramophone record (also phonograph record, or simply record) is an analogue sound storage medium consisting of a flat disc with an inscribed modulated spiral groove starting near the periphery and ending near the center of the disc.
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A phase vocoder is a type of vocoder which preserves both frequency and phase information.
A similar computer algorithm (referred to by the same name) allows frequency-domain modifications to a digital sound file (typically time expansion/compression and pitch shifting).
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A similar computer algorithm (referred to by the same name) allows frequency-domain modifications to a digital sound file (typically time expansion/compression and pitch shifting).
..... Click the link for more information.
A phase vocoder is a type of vocoder which preserves both frequency and phase information.
A similar computer algorithm (referred to by the same name) allows frequency-domain modifications to a digital sound file (typically time expansion/compression and pitch shifting).
..... Click the link for more information.
A similar computer algorithm (referred to by the same name) allows frequency-domain modifications to a digital sound file (typically time expansion/compression and pitch shifting).
..... Click the link for more information.
The short-time Fourier transform (STFT), or alternatively short-term Fourier transform, is a Fourier-related transform used to determine the sinusoidal frequency and phase content of local sections of a signal as it changes over time.
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discrete Fourier transform (DFT), occasionally called the finite Fourier transform, is a transform for Fourier analysis of finite-domain discrete-time signals. As with most Fourier analyses, it expresses an input function in terms of a sum of sinusoidal components by determining
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sine wave or sinusoid is a function that occurs often in mathematics, physics, signal processing, electrical engineering, and many other fields. Its most basic form is:
which describes a wavelike function of time (t) with
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which describes a wavelike function of time (t) with
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Smearing is a term used in rock climbing.
It is the practice of using the sole of a shoe against a flat rock face. Smearing can be one of the most insecure and technical techniques used in climbing, requiring a combination of leg/ankle tension, foot placement, and good
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It is the practice of using the sole of a shoe against a flat rock face. Smearing can be one of the most insecure and technical techniques used in climbing, requiring a combination of leg/ankle tension, foot placement, and good
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Lawrence R. Rabiner (born 28 September 1943 in Brooklyn, New York) is an electrical engineer working in the fields of digital signal processing and speech processing; in particular in digital signal processing for automatic speech recognition.
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Time domain is a term used to describe the analysis of mathematical functions, or physical signals, with respect to time. In the time domain, the signal or function's value is known for all real numbers, for the case of continuous time, or at various separate instants in the case
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In mathematics, a periodic function is a function that repeats its values after some definite period has been added to its independent variable.
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Examples
Everyday examples are seen when the variable is time..... Click the link for more information.
fundamental tone, often referred to simply as the fundamental and abbreviated fo, is the lowest frequency in a harmonic series.
The fundamental frequency (also called a natural frequency
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The fundamental frequency (also called a natural frequency
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A pitch detection algorithm (PDA) is an algorithm designed to estimate the pitch or fundamental frequency of a quasiperiodic or virtually periodic signal, usually a digital recording of speech or a musical note or tone. This can be done in the time domain or the frequency domain.
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Autocorrelation is a mathematical tool used frequently in signal processing for analysing functions or series of values, such as time domain signals. Informally, it is a measure of how well a signal matches a time-shifted version of itself, as a function of the amount of time shift.
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A cepstrum (pronounced /ˈkɛpstrəm/) is the result of taking the Fourier transform (FT) of the decibel spectrum as if it were a signal. Its name was derived by reversing the first four letters of "spectrum".
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In audio engineering, a fade is a gradual increase or decrease in the level of an audio signal.[1]
A recorded song may be gradually reduced to silence at its end (fade-out), or may gradually increase from silence at the beginning (fade-in
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A recorded song may be gradually reduced to silence at its end (fade-out), or may gradually increase from silence at the beginning (fade-in
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orchestra is an instrumental ensemble, usually fairly large with string, brass, woodwind sections, and possibly a percussion section as well. The term orchestra derives from the name for the area in front of an ancient Greek stage reserved for the Greek chorus.
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Adobe Audition (formerly Cool Edit Pro) is a digital audio editor computer program from Adobe Systems featuring both a multitrack, non-destructive mix/edit environment and a destructive-approach waveform editing view.
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hertz (symbol: Hz) is the SI unit of frequency. Its base unit is cycle/s or s-1 (also called inverse seconds, reciprocal seconds). In English, hertz is used as both singular and plural.
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Beats per minute (BPM) is a unit typically used as either a measure of tempo in music, or a measure of one's heart rate. A rate of 60 bpm means that one beat will occur every second. One bpm is equal to 1/60 Hz.
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A wavelet is a kind of mathematical function used to divide a given function into different frequency components and study each component with a resolution that matches its scale. A wavelet transform is the representation of a function by wavelets.
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In music transposition refers to the process of moving a collection of notes (pitches) up or down in pitch by a constant interval. For example, one might transpose an entire piece of music into another key.
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Pitch is the perceived fundamental frequency of a sound. While the actual fundamental frequency can be precisely determined through physical measurement, it may differ from the perceived pitch because of overtones, or partials, in the sound.
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Pitch shift is a sound recording technique, in which the normal pitch or tone of a sound is altered ("shifted"), for effect or for other purposes.
Pitch-shifting may be done both in analog and in digital recording.
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Pitch-shifting may be done both in analog and in digital recording.
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The mel scale, proposed by Stevens, Volkman and Newman in 1937 (J. Acoust. Soc. Am 8(3) 185--190) is a perceptual scale of pitches judged by listeners to be equal in distance from one another.
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In music theory, pitch spaces model relationships between pitches. These models typically use distance to model the degree of relatedness, with closely related pitches placed near one another, and less closely related pitches placed farther apart.
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