EP0285276A2 - Codage de formes d'ondes acoustiques - Google Patents

Codage de formes d'ondes acoustiques Download PDF

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Publication number
EP0285276A2
EP0285276A2 EP88302063A EP88302063A EP0285276A2 EP 0285276 A2 EP0285276 A2 EP 0285276A2 EP 88302063 A EP88302063 A EP 88302063A EP 88302063 A EP88302063 A EP 88302063A EP 0285276 A2 EP0285276 A2 EP 0285276A2
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European Patent Office
Prior art keywords
phase
coding
frequency components
frequency
frame
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EP88302063A
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German (de)
English (en)
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EP0285276B1 (fr
EP0285276A3 (en
Inventor
Robert J. Mcaulay
Thomas F. Quatieri, Jr.
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Massachusetts Institute of Technology
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Massachusetts Institute of Technology
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    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/02—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders

Definitions

  • the field of this invention is speech technology generally and, in particular, methods and devices for analyzing, digitally-encoding, modifying and synthesizing speech or other acoustic waveforms.
  • Digital speech coding methods and devices are the subject of considerable present interest, particularly at rates compatible with conventional transmission lines (i.e., 2.4 - 9.6 kilobits per second).
  • the typical approaches to speech modelling such as the so-called “binary excitation models” are ill-suited for coding applications and, even with linear predictive coding or other state of the art coding techniques, yield poor quality speech transmissions.
  • speech is viewed as the result of passing a glottal excitation waveform through a time-varying linear filter that models the resonant characteristics of the vocal tract. It is assumed that the glottal excitation can be in one of two possible states corresponding to voiced or unvoiced speech. In the voiced speech state the excitation is periodic with a period which varies slowly over time. In the unvoiced speech state, the glottal excitation is modeled as random noise with a flat spectrum.
  • Serial No. 712,866 discloses an alternative to the binary excitation model in which speech analysis and synthesis as well as coding can be accomplished simply and effectively by employing a time-frequency representation of the speech waveform which is independent of the speech state. Specifically, a sinusoidal model for the speech waveform is used to develop a new analysis-synthesis technique.
  • the basic method of U.S. Serial No. 712,866 includes the steps of: (a) selecting frames (i.e. windows of about 20 - 40 milliseconds) of samples from the waveform; (b) analyzing each frame of samples to extract a set of frequency components; (c) tracking the components from one frame to the next; and (d) interpolating the values of the components from one frame to the next to obtain a parametric representation of the waveform.
  • a synthetic waveform can then be constructed by generating a set of sine waves corresponding to the parametric representation.
  • the method is employed to choose amplitudes, frequencies, and phases corresponding to the largest peaks in a periodogram of the measured signal, independently of the speech state.
  • the amplitudes, frequencies, and phases of the sine waves estimated on one frame are matched and allowed to continuously evolve into the corresponding parameter set on the successive frame. Because the number of estimated peaks is not constant and is slowly varying, the matching process is not straightforward. Rapidly varying regions of speech such as unvoiced/voiced transitions can result in large changes in both the location and number of peaks.
  • phase continuity of each sinusoidal component is ensured by unwrapping the phase.
  • the phase is unwrapped using a cubic phase interpolation function having parameter values that are chosen to satisfy the measured phase and frequency constraints at the frame boundaries while maintaining maximal smoothness over the frame duration.
  • the corresponding sinusoidal amplitudes are simply interpolated in a linear manner across each frame.
  • pitch estimates can be used to establish a set of harmonic frequency bins to which the frequency components are assigned.
  • Pitch is used herein to mean the fundamental rate at which a speaker's vocal cords are vibrating).
  • the amplitudes of the components are coded directly using adaptive differential pulse code modulation (ADPCM) across frequency or indirectly using linear predictive coding.
  • ADPCM adaptive differential pulse code modulation
  • the peak having the largest amplitude is selected and assigned to the frequency at the center of the bin. This results in a harmonic series based upon the coded pitch period.
  • the phases are then coded by using the frequencies to predict phase at the end of the frame, unwrapping the measured phase with respect to this prediction and then coding the phase residual using 4-5 bits per phase peak.
  • New encoding techniques based on a sinusoidal speech representation model are disclosed.
  • a pitch-adaptive channel encoding technique for amplitude coding is disclosed in which the channel spacing is varied in accordance with the pitch of the speaker's voice.
  • a phase synthesis technique is disclosed which locks rapidly-varying phases into synchrony with the phase of the fundamental.
  • the parameters of the sinusoidal model are the amplitudes, frequencies and phases of the underlying sine waves, and since for a typical low-pitched speaker there can be as many as 80 sine waves in a 4 kHz speech bandwidth, it is not possible to code all of the parameters directly and achieve transmission rates below 9.6 kbps.
  • the first step in reducing the size of the parameter set to be coded is to employ a pitch extraction algorithm which lead to a harmonic set of sine waves that are a "perceptual" best fit to the measured sine waves.
  • a pitch extraction algorithm which lead to a harmonic set of sine waves that are a "perceptual" best fit to the measured sine waves.
  • a predictive model for the phases of the sine waves is also developed, which not only leads to a set of residual phases whose dynamic ranges are a fraction of the [- ⁇ , ⁇ ] extent of the measured phases, but also leads to a model from which the phases of the high frequency, sine waves can be regenerated from the set of coded baseband phases.
  • very natural and intelligible coded speech is obtained at 8.0 kbps.
  • STC Sinusoidal Transform Coder
  • DPCM differential pulse code modulation
  • a set of linearly-spaced frequencies in the baseband and a further set of logarithmically-space frequencies in the higher frequency region are employed in the transmitter to code amplitudes.
  • another amplitude envelope is constructed by linearly interpolating between the channel amplitudes. This is then sampled at the pitch harmonics to produce the set of sine-wave amplitudes to be used for synthesis.
  • the system phase can be predicted from the coded log-amplitude using homomorphic techniques which when combined with a prediction of the excitation phase can restore complete fidelity during synthesis by merely coding phase residuals.
  • phase predictions are poor, but the same sort of behavior can be simulated by replacing each residual phase by a uniformly-distributed random variable whose standard deviation is proportional to the degree to which the analyzed speech is unvoiced.
  • a coding scheme for a very low data rate transmission lines (i.e., below 4.8 kbps), a coding scheme has been devised that essentially eliminates the need to code phase information.
  • systems are disclosed herein for maintaining phase coherence and introducting an artificial phase dispersion.
  • a synthetic phase model is disclosed which phase-locks all the sine waves to the fundamental and adds a pitch-dependent quadratic phase dispersion and a voicing-dependent random phase to each phase track.
  • Speech is analyzed herein as having two components to the phase: a rapidly-varying component that changes with every sample and a slowly varying component that changes with every frame.
  • the rapidly-varying phases are locked into synchrony with the phase of the fundamental and, furthermore, the pitch onset time simply establishes the time at which all the excitation sine waves come into phase. Since the sine waves are phase-locked, this onset time represents a delay which is not perceptible by the ear and, hence, can be ignored. Therefore, the phase of the fundamental can be generated by integrating the instantaneous pitch frequency and the rapidly-varying phases will be multiples of the phase of the fundamental.
  • the speech waveform is modeled as a sum of sine waves.
  • the first step in coding speech is to express the input speech waveform, s(n), in terms of the sinusoidal model, where A k , ⁇ k and ⁇ k are the amplitudes, frequencies and phases corresponding to the peaks of the magnitude of the high-resolution short-time Fourier transform. It should be noted that the measured frequencies will not in general be harmonic.
  • the speech waveform can be modeled as the result of passing a glottal excitation waveform through a vocal tract filter.
  • (3a) ⁇ k ⁇ k - arg H( ⁇ k ). (3b)
  • a k A k /
  • ⁇ k ⁇ k - arg H( ⁇ k ).
  • 3b) In order to calculate the excitation phase in (3b), it is necessary to compute the amplitude and phase of the vocal tract filter. This can be done either by using homomorphic techniques or by fitting an all-pole model to the measured sine-wave amplitudes. These techniques are discussed in U.S. Serial No. 712,866.
  • FIG. 1 is a block diagram showing the basic analysis/synthesis system of the present invention.
  • the peaks of the magnitude of the discrete Fourier transform (DFT) of a windowed waveform are found simply by determining the locations of a change in slope (concave down).
  • Phase measurements are derived from the discrete Fourier transform by computing the arctangents at the estimated frequency peaks.
  • the speech waveform can be digitized at a 10kHz sampling rate, low-passed filtered at 5 kHz, and analyzed at 10-20 msec frame intervals employing an analysis window of variable duration in which the width of the analysis window is pitched adaptive, being set, for example, at 2.5 times the average pitch period with a minimum width of 20 msec.
  • STC sinusoidal transform coder
  • One way to encode amplitude information at low rates is to exploit a perception-based strategy.
  • DPCM DPCM
  • further efficiencies are gained by allowing the channel separation to increase logarithmically with frequency, thereby exploiting the critical band properties for the ear. This can be done by constructing an envelope of the sine-wave amplitudes by linearly interpolating between sine-wave peaks. This envelope is then sampled at predefined frequencies.
  • a 22-channel design was developed which allowed for 9 linearly-spaced frequencies at 93 Hz/channel in the baseband and 11 logarithmically-spaced frequencies in the higher-frequency region.
  • DPCM coding was used with 3 bits/channel for the channels 2 to 9 and 2 bits/channel for channels 10 to 22. It is not necessary to explicitly code channel 1 since its level is chosen to obtain the desired energy level.
  • another amplitude envelope is constructed by lineary interpolating between the channel amplitudes. This is then sampled at the pitch harmonics to produce the set of sine-wave amplitudes to be used for synthesis.
  • the expansion factor ⁇ is chosen such that F N is close to the 4000 Hz band edge. If the pitch is at or below 93 Hz, then the fixed 93 Hz linear/logarithmic design can be used, and if it is above 93 Hz, then the pitch-adaptive linear/log design can be used. Furthermore, if the pitch is above 174 Hz, then a strictly linear design can be used. In addition, the bit allocation per channel can be pitch-adaptive to make efficient use of all of the available bits.
  • the DPCM encoder is then Applied to the logarithm of the envelope samples at the pitch-adaptive channel frequencies. Since the quantization noise has essentially a flat spectrum in the quefrequency domain (the Fourier transform of the log magnitudes) and since the speech envelope spectrum varies as 1/n2 in this domain, then optimal reduction of the quantization noise is possible by designing a Weiner filter. This can be approximated by an appropriately designed cepstral low-pass filter.
  • This amplitude encoding algorithm was implemented on a real-time facility and evaluated using the Diagnostic Rhyme Test. For 3 male speakers, the average scores were 95.2 in the quiet, 92.5 in airborne-command-post noise and 92.2 in office noise. For females, the scores were about 2 DRT points lower in each case.
  • the pitch-adaptive 22-channel amplitude encoder is designed for operation at 4.8 kbps, it can operate at any rate from 1.8 kbps to 8.0 kbps simply by changing the bit allocations for the amplitudes and phases. Operation at rates below 4.8 kbps was most easily obtained by eliminating the phase coding. This effectively defaulted the coder into a "magnitude-only" analysis/synthesis system whereby the phase tracks are obtained simply by integrating the instantaneous frequencies associated with each of the sine waves. In this way, operation at 3.1 kbps was achieved without any modification to the amplitude encoder. By further reducing the bit allocations for each channel, operation at rates down to 1.8 kbps was possible.
  • phase modeling is to develop a parametric model to describe the phase measurements in (4).
  • the intuition behind the new phase model stems from the fact that during steady voicing the excitation waveform will consist of a sequence of pitch pulses.
  • a pitch pulse occurs when all of the sine waves add coherently (i.e., are in phase).
  • n o is the onset time of the pitch pulse measured with respect to the center of the analysis frame.
  • the phase model depends on the two parameters, n o and ⁇ which should be chosen to make e(n) "close to" e(n).
  • n o denotes the maximizing value
  • the function l(n o ) is highly non-linear in n o , and it is not possible to find a simple analytical solution for the optimum value.
  • the first step used in coding the sine wave parameters is to assign one sine wave to each harmonic frequency bin. Since it is this set of sine wave which will ultimately be reconstructed at the receiver, it is to this reduced set of sine waves that the new phase model will be applied.
  • an amplitude envelope is created by applying linear interpolation to the amplitudes of the reduced set of sine waves. This is used to flatten the amplitudes and then homomorphic methods are used to estimate and remove the system phase to create the sine wave representation of the glottal excitation waveform. The onset time and the system phase ambiguity are then estimated and used to form a set of residual phases. If the model were perfect, then these phase residuals would be zero.
  • the model is not perfect; hence, for good synthetic speech it is necessary to code the residuals.
  • An example of such a set of residuals is shown in FIG. 4 for the same data illustrated in FIG 2. Since only the sine waves in the baseband (up to 1000 Hz) will be coded, the model is actually fitted to the sine wave phase data only in the baseband region. The main point is that whereas the original phase measurements has values that were uniformly distributed over the [- ⁇ , ⁇ ) region, the dynamic range of the phase residuals is much less than ⁇ , hence, coding efficiencies can be obtained.
  • the final step in coding the sine wave parameters is to quantize the frequencies. This is done by quantizing the residual frequency obtained by replacing the measured frequency by the center frequency of the harmonic bin in which the sine wave lies. Because of the close relationship between the measured excitation phase of a sine wave and its frequency, it is desirable to compensate the phase should the quantized frequency be significantly different from the measured value. Since the final decoded excitation phase is the phase predicted by the model plus the coded phase residual, some phase compensation is inherent in the process since the phase model will be evaluated at the coded frequency and, hence, will better preserve the pitch structure in the synthetic waveform.
  • the glottal excitation can be thought of as a sequence of periodic impulses which can be decomposed into a set of harmonic sine waves that add coherently at the time of occurrence of each pitch pulse.
  • A( ⁇ ) is the amplitude envelope
  • n o is the pitch onset time
  • ⁇ o is the pitch frequency
  • ⁇ ( ⁇ ) is the system phase
  • ⁇ (m ⁇ o ) is the residual phase at the m th harmonic
  • ⁇ 2 ⁇ f/f s is the angular frequency in radians, relative to the sampling frequency f s . Since under a minimum-phase assumption the system phase can be determined from the coded log-amplitude using homomorphic techniques, then the fidelity of the harmonic reconstruction depends only on the number of bits that can be assigned to the coding of the phase residuals.
  • phase residuals that were essentially zero, while during unvoiced speech, the phase predictions were poor resulting in phase residuals that appeared to be random values within [- ⁇ , ⁇ ].
  • the behavior of the phase residuals was somewhere between these two extremes.
  • the same sort of behavior can be simulated by replacing each residual phase by a uniformly-distributed random variable whose standard deviation is proportional to the degree to which the analyzed speech is unvoiced.
  • phase dispersion Since the system phase ⁇ ( ⁇ ) is derived from the coded log-magnitude, it is minimum-phase, which causes the synthetic waveform to be "spiky" and, in turn, leads to the perceived "buzziness".
  • the flexibility of the STC system allows for a pitch-adaptive speaker-dependent design.
  • phase model There are two components to the phase: a rapidly-varying component that changes with every sample, and a slowly-varying component that changes with every frame.
  • phase-locked synthesizer has been implemented on the real-time system and found to dramatically improve the quality of the synthetic speech. Although the improvements are most noticeable at the lower rates below 3 kbps where no phase coding is possible, the phase-locking technique can also be used for high-frequency regeneration in those cases where not all of the baseband phases are coded. In fact, very good quality can be obtained at 4.8 kbps while coding fewer phases than was used in the earlier designs. Furthermore, since Eqs. (16-20) depend only on the measured pitch frequency, ⁇ o , and a voicing probability, P v , reduction in the data rate below 4.8 kbps is not possible with less loss in quality even though no explicit phase information is coded.
EP88302063A 1987-04-02 1988-03-10 Codage de formes d'ondes acoustiques Expired - Lifetime EP0285276B1 (fr)

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Application Number Priority Date Filing Date Title
AT88302063T ATE95936T1 (de) 1987-04-02 1988-03-10 Codierung von akustischen wellenformen.

Applications Claiming Priority (2)

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US3409787A 1987-04-02 1987-04-02
US34097 1998-03-03

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EP0285276A2 true EP0285276A2 (fr) 1988-10-05
EP0285276A3 EP0285276A3 (en) 1989-11-23
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JP (2) JP3191926B2 (fr)
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CA (1) CA1332982C (fr)
DE (1) DE3884839T2 (fr)

Cited By (9)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US5029509A (en) * 1989-05-10 1991-07-09 Board Of Trustees Of The Leland Stanford Junior University Musical synthesizer combining deterministic and stochastic waveforms
EP0527535A2 (fr) * 1991-08-14 1993-02-17 Philips Patentverwaltung GmbH Dispositif pour la transmission du langage
US5327521A (en) * 1992-03-02 1994-07-05 The Walt Disney Company Speech transformation system
EP0642129A1 (fr) * 1993-08-02 1995-03-08 Koninklijke Philips Electronics N.V. Système de transmission avec reconstruction des échantillons manquants
EP0666557A2 (fr) * 1994-02-08 1995-08-09 AT&T Corp. Interpolation de formes d'onde par décomposition en bruit et en signaux périodiques
EP0780831A2 (fr) * 1995-12-23 1997-06-25 Nec Corporation Procédé de codage de la parole ou de la musique avec quantification des composants harmoniques en particulier et des composants résiduels par la suite
EP1008138A1 (fr) * 1996-11-07 2000-06-14 Creative Technology Ltd. Systeme de modification de signaux audio par utilisation de transformees de fourier
WO2002003381A1 (fr) * 2000-02-29 2002-01-10 Qualcomm Incorporated Procede et appareil pour mesurer la phase d'un signal quasi periodique
US6449592B1 (en) 1999-02-26 2002-09-10 Qualcomm Incorporated Method and apparatus for tracking the phase of a quasi-periodic signal

Families Citing this family (7)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JPH04150233A (ja) * 1990-10-09 1992-05-22 Matsushita Electric Ind Co Ltd 信号伝送方法
JP2606756B2 (ja) * 1990-10-22 1997-05-07 財団法人鉄道総合技術研究所 ディジタル信号伝送装置
DE60113034T2 (de) * 2000-06-20 2006-06-14 Koninkl Philips Electronics Nv Sinusoidale kodierung
AU2003274617A1 (en) * 2002-11-29 2004-06-23 Koninklijke Philips Electronics N.V. Audio coding
WO2005024783A1 (fr) * 2003-09-05 2005-03-17 Koninklijke Philips Electronics N.V. Codage audio a faible debit binaire
US8805694B2 (en) 2009-02-16 2014-08-12 Electronics And Telecommunications Research Institute Method and apparatus for encoding and decoding audio signal using adaptive sinusoidal coding
CN102264022B (zh) * 2010-04-08 2014-03-12 Gn瑞声达公司 助听器的稳定性改进

Citations (1)

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Publication number Priority date Publication date Assignee Title
WO1986005617A1 (fr) * 1985-03-18 1986-09-25 Massachusetts Institute Of Technology Traitement de formes d'ondes acoustiques

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US4856068A (en) * 1985-03-18 1989-08-08 Massachusetts Institute Of Technology Audio pre-processing methods and apparatus

Patent Citations (1)

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Publication number Priority date Publication date Assignee Title
WO1986005617A1 (fr) * 1985-03-18 1986-09-25 Massachusetts Institute Of Technology Traitement de formes d'ondes acoustiques

Non-Patent Citations (3)

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Title
ICASSP 85 PROCEEDINGS IEEE INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH, AND SIGNAL PROCESSING, Tampa, 26th-29th March 1985, vol. 2, pages 489-492, IEEE; T.E. QUATIERI et al.: "Speech transformations based on a sinusoidal representation" *
ICASSP 85 PROCEEDINGS IEEE INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH, AND SIGNAL PROCESSING, Tampa, 26th-29th March 1985, vol. 3, pages 945-948, IEEE; R.J. McAULAY et al.: "Mid-rate coding based on a sinusoidal representation of speech" *
ICASSP 87 PROCEEDINGS IEEE INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH, AND SIGNAL PROCESSING, Dallas, 6th-9th April 1987, vol. 3, pages 1645-1648, IEEE; R.J. McAULAY et al.: "Multirate sinusoidal transform coding at rates from 2.4 KBPS to 8 KBPS" *

Cited By (14)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US5029509A (en) * 1989-05-10 1991-07-09 Board Of Trustees Of The Leland Stanford Junior University Musical synthesizer combining deterministic and stochastic waveforms
EP0527535A2 (fr) * 1991-08-14 1993-02-17 Philips Patentverwaltung GmbH Dispositif pour la transmission du langage
EP0527535A3 (en) * 1991-08-14 1993-10-20 Philips Patentverwaltung Apparatus for transmission of speech
US5327521A (en) * 1992-03-02 1994-07-05 The Walt Disney Company Speech transformation system
EP0642129A1 (fr) * 1993-08-02 1995-03-08 Koninklijke Philips Electronics N.V. Système de transmission avec reconstruction des échantillons manquants
BE1007428A3 (nl) * 1993-08-02 1995-06-13 Philips Electronics Nv Transmissiesysteem met reconstructie van ontbrekende signaalmonsters.
EP0666557A2 (fr) * 1994-02-08 1995-08-09 AT&T Corp. Interpolation de formes d'onde par décomposition en bruit et en signaux périodiques
EP0666557A3 (fr) * 1994-02-08 1997-08-06 At & T Corp Interpolation de formes d'onde par décomposition en bruit et en signaux périodiques.
EP0780831A2 (fr) * 1995-12-23 1997-06-25 Nec Corporation Procédé de codage de la parole ou de la musique avec quantification des composants harmoniques en particulier et des composants résiduels par la suite
EP0780831A3 (fr) * 1995-12-23 1998-08-05 Nec Corporation Procédé de codage de la parole ou de la musique avec quantification des composants harmoniques en particulier et des composants résiduels par la suite
EP1008138A1 (fr) * 1996-11-07 2000-06-14 Creative Technology Ltd. Systeme de modification de signaux audio par utilisation de transformees de fourier
EP1008138A4 (fr) * 1996-11-07 2002-02-20 Creative Technoloy Ltd Systeme de modification de signaux audio par utilisation de transformees de fourier
US6449592B1 (en) 1999-02-26 2002-09-10 Qualcomm Incorporated Method and apparatus for tracking the phase of a quasi-periodic signal
WO2002003381A1 (fr) * 2000-02-29 2002-01-10 Qualcomm Incorporated Procede et appareil pour mesurer la phase d'un signal quasi periodique

Also Published As

Publication number Publication date
EP0285276B1 (fr) 1993-10-13
AU7436491A (en) 1991-07-11
CA1332982C (fr) 1994-11-08
AU1314588A (en) 1988-10-06
AU643769B2 (en) 1993-11-25
AU612351B2 (en) 1991-07-11
ATE95936T1 (de) 1993-10-15
DE3884839D1 (de) 1993-11-18
JP2001228898A (ja) 2001-08-24
DE3884839T2 (de) 1994-05-05
JP3191926B2 (ja) 2001-07-23
EP0285276A3 (en) 1989-11-23
JPH01221800A (ja) 1989-09-05

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