Formant Synthesis of Haegeum: A Sound Analysis/Synthesis System using Cpestral Envelope
|
|
- Harold Ross
- 5 years ago
- Views:
Transcription
1 Formant Synthesis of Haegeum: A Sound Analysis/Synthesis System using Cpestral Envelope Myeongsu Kang School of Computer Engineering and Information Technology Ulsan, South Korea ilmareboy@ulsan.ac.kr Yeonwoo Hong 1 School of Electrical Engineering Ulsan, South Korea ducj16@ulsan.ac.kr Abstract This paper presents a formant synthesis method of haegeum using cepstral envelope for spectral modeling. Spectral modeling synthesis (SMS) is a technique that models timevarying spectra as a combination of sinusoids (the "deterministic" part), and a time-varying filtered noise component (the "stochastic" part). SMS is appropriate for synthesizing sounds of string and wind instruments whose harmonics are evenly distributed over whole frequency band. Formants (or acoustic resonances) are extracted from cepstral envelope and we use them for synthesizing sinusoids. A secondorder digital resonator by the impulse-invariant transform (IIT) is applied to generate deterministic components and the results are band-pass filtered to adjust magnitude. The noise is calculated by first generating the sinusoids with formant synthesis, subtracting them from the original sound, and then removing some harmonics remained. A line-segment approximation is used to model noise components. The synthesized sounds are consequently by adding sinusoids, which are shown to be similar to the original haegeum sounds. Keywords-Sound synthesis of Haegeum, spectral modeling, formant synthesis, cepstral envelope, spectral analysis I. INTRODUCTION The study on synthesis of musical instrumental sounds has been studied through the method of sampling, modulation, filtering, modeling, etc [1]. Sampling and filtering are the most traditional methods that use recorded instrumental sounds or produce a desired waveform and spectrum using filters. We can synthesize the most natural sound through sampling that well describes the sound color of the original instruments. But sampling is appropriate for the solo play and we have to resample when the playing style is changed. Using the modulation we can create new sounds such as electrical sounds not natural sounds. The synthesized sounds using modulation feels like artificial, so it's very hard to produce natural sounds of various instruments. On the other hands, modeling is a method that synthesizes the musical sounds using digital filters designed based on acoustical characteristics. When using modeling we can adjust the tone color by changing the filter parameters and can describe the playing style by adding or deleting certain filters. But the calculation complexity is much higher [2]. To reproduce realistic instrumental sounds it is important to select a good model. The synthesis techniques of musical instruments using modeling are physical modeling and spectral modeling. Physical modeling is synthesis techniques that analyze the sound production mechanism and design appropriate model. Physical modeling using digital wave guide is much used in the area of sound synthesis. Spectral modeling synthesizes the sounds by analyzing the spectrum of the instrumental sounds as a sum of sinusoids and other components which affect the characteristic of the instrumental sounds. Spectral modeling is appropriate for the synthesis of string and wind instruments which have periodic character [3]. The main advantage of spectral modeling is the existence of analysis procedures that extract the synthesis parameters out of real sounds, thus being able to reproduce actual sounds [4]. Additive synthesis is the original spectral modeling technique. Generally, the techniques which models instrumental sounds as a sum of sinusoids and noise components are referred to as spectral modeling [5]. The modeling of sinusoids is more important because noise components are not much related to the pitch. Additive synthesis, subtractive synthesis, formant synthesis can be used for the modeling of sinusoids. Formant synthesis is a technique which synthesizes harmonics using spectral envelope in the frequency domain. The information for magnitudes and frequencies of every harmonics is not needed, and the inharmonic components can be synthesized using formant synthesis [6]. To extract spectral envelope, the followings are generally used for formant synthesis: interpolation, linear prediction coding (LPC), and cepstrum. In this paper, cepstrum is used. In 1963, Bogert et al. introduced cepstral processing which had been studied for seismic analysis. At about same time, Oppenheim proposed a new class of systems called homomorphic system whose fundamental concept is same as cepstrum [7]. Cepstral coefficients are extracted from Fourier transform of logarithm of input spectrum. Input signal combined by convolution in time domain is converted into a form of product in frequency domain, and converted into a summation by logarithm. We can interpret Fourier transform of the logarithm as a superposition of the components included in input signal in summed form. In 1990, Serra introduced spectral modeling synthesis (SMS) that synthesizes musical 1 Corresponding author. This research was supported by Basic Science Research Program through the National Research Foundation of Korea (NRF) funded by the Ministry of Education, Science and Technology ( ) /11/$ IEEE
2 instrumental sounds using spectral modeling technique [4]. After that, sound synthesis of musical instrument using SMS has been studied [3, 8]. Sinusoids+noise model is developed as sinusoids+noise+transient model [9 11]. Sound synthesis of Korean traditional instruments using spectral modeling has not yet studied. A study on analysis of acoustical characteristics of Korean traditional bowed string instrument, haegeum, has done [12]. However, a study on synthesis of haegeum sounds has not yet done. Haegeum has a characteristic that harmonics are evenly distributed through very high frequency band, similar to wood wind instruments. Consequently, spectral modeling is suitable for haegeum. In this paper, we propose a formant synthesis techniques using cepstral envelope for the spectral modeling of haegeum. Formants are extracted from cepstral envelope and utilized as parameters for the synthesis of sinusoids. The filter by the impulse-invariant transform (IIT) is used as a digital resonator for the synthesis of sinusoids. For the interpolation of the magnitude of filtered signal, band-pass filter is applied to the output of the resonator. The noise is calculated by first generating the sinusoids with formant synthesis, subtracting them from the original sounds. As a result, the synthesized single-notes are produced by adding sinusoids by formant synthesis and noise components by a linesegment approximation. The rest of this paper is organized as follows. Section II introduces the structure of haegeum and Section III gives background about the cepstrum. Then, Section IV shows frequency characteristics of haegeum and explains how to extract spectral synthesis parameters. The synthetic results are illustrated in Section V, and discussions are mentioned in Section IV. Section IIV concludes this paper. II. A BOWED INSTRUMENT: HAEGEUM A. Structure of Haegeum Haegeum is a Korean traditional bowed string instrument that produces sound by rubbing two strings with a bow. Haegeum has characteristics of both wind and string instruments. Consequently, it can be defined a wind instrument by its structure and a string instrument due to its two strings. As shown in Fig. 1, the thick and thin strings are called Junghyun and Yuhyun, respectively. The other parts of the haegeum include: the cylindrical body, the Jua, the Bokpan, the Wonsan, the Ipjuk, and the fiddle bow. The Jua tightens the two strings. One side of the cylindrical body is hollow; the other is flat and is known as the Bokpan. Sound echoes though the Bokpan. The Wonsan elevates the strings above the body and various sounds are produced based on its position. The Ipjuk functions as a handle and neck. The fiddle bow is made from horsehair and it is always inserted between the two strings attached to a small barrel-shaped resonator (Bokpan). The two strings are generally tuned to A3 and E4 and the resonator is tuned to about 300Hz. The resonator has a paulownia wood plate at one end on which the bridge rests. B. Musical Range on Haegeum The pitch of the haegeum changes with the position of lefthand on the strings and it is usually played at the following three positions: Hwangjong (near the Jua), Jungryeo (middle of the strings), and Chonghwangjong (near the Bokpan). The haegeum has a wide range of two and a half octaves as shown in Table I. As you can see, there are some duplicated singlenotes at different positions. (e.g., note Joong (Ab4) at Hwangjong and Jungryeo position). TABLE I. Hwangjong position SINGLE-NOTES AT DIFFERENT POSITIONS Joong (Ab3) 210 Im (Bb3) 228 Nam (C4) 257 Hwang (Eb4) 317 Tae (F4) 354 Joong (Ab4) 433 Im (Bb4) 478 Nam (C5) 524 Jungryeo position Hwang (Eb4) 315 Tae (F4) 351 Joong (Ab4) 427 Im (Bb4) 474 Nam (C5) 527 Mu (Db5) 556 Hwang (Eb5) 611 Tae (F5) 668 Cheonghwangjong position Joong (Ab4) 408 Im (Bb4) 457 Nam (C5) 519 Hwang (Eb5) 608 Tae (F5) 687 Joong (Ab5) 827 Im (Bb5) 930 Nam (C6) 1013 Figure 1. Structure of a Korean traditional bowed string instrument, Haegeum The term yul corresponds to the term note in Western music. In Korean musical theory, an octave is divided into twelve tones and the tones are named as follows: Hwangjong
3 (Eb), Daeryo (E), Taeju (F), Hyeopjeong (Gb), Goseon (G), Jungryeo (Ab), Yubin (A), Imjong (Ab), Ichik (B), Namryeo (C), Muyeok (Db), and Eungjong (D). The bolded syllables are those in current usage in notation for haegeum. sustain. Constant vibrational energy is delivered to the haegeum sound and the sound only fades once the bowing stops. The haegeum generates considerably regular harmonics and they are distributed to 20 khz. III. BACKGROUND: CEPSTRUM It is convenient to assume that the input signal convolved with an excitation signal and a filter can be expressed as follow xn ( ) = en ( ) sn ( ), (1) where x(n) is the input signal, e(n) is the excitation signal and s(n) is a kind of filter that generates the input signal by filtering the given excitation signal. Taking the logarithm of the Fourier transform of both side of (1) yields jω jω jω log X( e ) = log E( e ) + log S( e ). (2) ω E yields a spectrum that should be characterized by a relatively rapidly varying function of ω j The term ( e ) j (e.g., a noise signal), the term ( e ) S ω varies more slowly with ω (e.g., harmonic components of musical sound). Consequently, the left-hand side of (2) can be separated into the two right-hand-side components by a kind of a filter that separates the log spectral components that vary rapidly with ω from those that vary slowly with ω. The cepstrum is calculated by taking the inverse discrete Fourier transform of the left-hand side of (2) yielding Figure 2. Comparison of waveforms: (a) gayageum and (b) haegeum Furthermore, Fig. 3 presents the maximum amplitude is in the range from 1600Hz to 1800Hz. This frequency characteristic is the same as the frequency characteristic of the body of the haegeum in [12]. This concludes that the body gives the haegeum its distinctive tone. As shown in Fig. 3, several large amplitudes are appeared out of 1600Hz 1800Hz and these formants correspond to resonant frequencies that affect the characteristic of the haegeum sound. 1 π jω jωn cn ( ) = π log Xe ( ) e dω, 2 (3) where c(n) is called the nth cepstral coefficient [14]. The spectral envelope, which varies slowly with respect to frequency, yields large-valued cepstral coefficients for low values of n, but it dies out for high n. The spectral fine structure is rapidly varying with ω, and it yields small-valued cepstral coefficients for small n. Consequently, the contribution of the excitation and the filter can be separated in the cepstral domain by taking the Fourier transform of the cepstral coefficients: sinusoidal component and stochastic component (e.g., excitation signal). IV. EXTRACTION OF SYNTHESIS PARAMETERS FOR HAEGEUM Figure 3. Formants (or acoustical resonances) of note Hwang (Eb4) at Hwangjong position A. Frequency Characteristics of Haegeum The haegeum produces a variety of sounds and it can generate very sharp sounds when the length of a string is reduced by pressing a finger down on it. Moreover, a player produces sound by pushing/drawing a bow across one or more strings, and thus the produced haegeum sounds can be characterized by the bow velocity. Fig. 2 shows the waveform comparison between haegeum and gayageum, which is a Korean traditional plucked-string instrument. The haegeum has a slightly slower attack than the gayageum. Since the haegeum is bowed to generate sound, there is no decay and a very long B. Extraction of Spectral Parameters As mentioned in Section II, there are three different positions to play the haegeum. For haegeum sounds generated in each position, several single-notes are duplicated. With this reason, single-notes generated at Hwangjong position are considered to extract spectral parameters in this paper. Spectral parameters are extracted from the sustain region because the haegeum has a slower attack and no decay. In this case, it can be assumed that the steady bow velocity is supplied by the player and thus it is not necessary to consider about the sinusoidal frequency and phase content of local sections of a
4 Figure 4. A flow diagram to extract spectral synthesis parameter signal as it changes over time. Therefore, we do not consider about the change of harmonics and their magnitudes over time through the short-time Fourier transform (STFT) in this paper. The fast Fourier transform (FFT) is performed for signal of 500 1,000 samples size extracted from the sustain region of each note as a frame of STFT analysis. Fig. 4 illustrates a flow diagram to extract spectral parameters to analyze the haegeum sound. In formant synthesis, the basic assumption is that the transfer function can be satisfactorily modeled by simulating formant (or resonant) frequencies, formant amplitudes and bandwidths. Thus, the synthesis consists of the artificial reconstruction of the formant characteristics to be produced. This is done by exciting a set of resonators to achieve the desired sound spectrum. To extract these synthesis parameters, it is very important to determine an appropriate window size to truncate cepstral coefficients. Fig. 5 shows the real cepstrum of note Hwang (Eb4) and the window size is set to a period in length. We can finally obtain the cepstral envelope from the truncated cepstrum by taking Fourier transform. the following two conditions as efficient resonances to well represent frequency characteristics of the haegeum sound: i) the interval of two adjacent resonances should be greater than the fundamental frequency, and ii) the amplitude of the current resonance should be bigger than the amplitude of adjacent resonances. The bandwidth of each resonance can be measured at half-power points (gain -3dB or relative to peak). Fig. 6 illustrates the cepstral envelope and extracted resonances from note Im at Hwangjong position. Figure 6. Cepstral envelope and acoustical resonances of note Im at Hwangjong position Figure 5. Real cepstrum of note Hwang (solid) and window to extract cepstral envelope (dotted) The cepstral envelope represents the formant structure and many resonances lie within 20 khz. Although many resonances lie within 20 khz, the amplitude of resonance dramatically goes down at above 15 khz as shown in Fig. 3. Consequently, we consider resonances within 15 khz that are satisfied with As a result, sinusoids are synthesized with these extracted resonances and then it is possible to obtain noise components (or residual signal) by subtracting the synthesized sinusoids from the original sound. Furthermore, it is possible to know how well the extracted resonances are matched with the harmonic components of the original sound. If sinusoids remain in the residual signal, we should reanalyze the sound until we obtain a good enough residual signal that is free of sinusoidal components. Ideally the resulting residual signal must be as close as possible to a stochastic signal. To model the residual signal, we should firstly obtain the local maximum values at every 100 samples in the original residual, and then a line-segment approximation is applied to its log-magnitude spectrum.
5 V. SYNTHESIS OF HAEGEUM SOUNDS A digital resonator generating the sinusoids is illustrated in Fig. 7 and two parameters are used to specify the input-output characteristics of the resonator: the resonant (or formant) frequency, f r, and the resonance bandwidth ω BW. resonant frequencies are 250Hz and 550Hz, the center frequency, bandwidth, upper and lower cutoff frequencies of the BSF should be 250Hz, 300Hz, 100Hz and 400Hz, respectively. Fig. 8 depicts spectra of synthetic notes Im (Bb4) and Nam (C5) by band-pass filtered formant synthesis. Figure 8. Spectra of synthesized signle-notes by formant synthesis applied with a digital band-pass filter: (a) Im (Bb4) and (b) Nam (C5) Figure 7. Frequency response of resonator (f r = 1000Hz, ω BW = 50Hz) Then, samples of the output of a digital resonator, y(n), are computed from the input signal, x(n), by (4) yn ( ) = Axn ( ) + Byn ( 1) + Cyn ( 2). (4) The constants A, B and C are related to the resonant frequency f r and the bandwidth ω BW by the impulse-invariant transformation. ωbw C = e ωbw B = 2e 2 cos( ωr ) (5) A = 1 B C. The digital resonator is a second-order difference equation and the transfer function of the digital resonator is given by (6) j e ω A T( z) = 1 2, 1 Bz Cz (6) where z = [6]. To produce the sinusoids, a parallel formant synthesizer is used in this paper. The parallel formant synthesizer sums the outputs of the simultaneously excited formant resonators. In this case, a superposition of resonators at adjacent resonant frequencies can be occurred and this may cause the result that attenuation around resonances is not described well. To conquer this drawback, a digital band-pass filter (BSF) is connected to the output of each formant resonator. The center frequency and bandwidth are specified by each resonant frequency and the difference of two adjacent resonant frequencies, respectively. If, for example, the first and second The synthetic spectra are not satisfied as much as we expected. This results from the lack of stochastic components. As mentioned in previous section, the line-segment approximation is applied for modeling noise components, and parameters are specified by the local maxima at every 100 samples in the residual. Fig. 9 shows a comparison of original noise components and synthesized noise components by the line-segment approximation. Figure 9. A comparison of original noise components (dotted) and synthesized noise components (solid) of Nam (C5) by the line-segment approximation. Fig. 10 illustrates a block diagram to synthesize single-notes of haegeum. As a result, the system includes digital resonators, digital band-pass filters, and the line-segment approximation.
6 Figure 10. A block diagram to synthesize single-notes of haegeum with extracted spectral parameters Figure 11. A comparison of spectra between original single-notes (dotted) and synthesized single-notes (solid) at Hwangjong position: (a) Joong (Ab3), (b) Im (Bb3), (c) Nam (C4), (d) Hwang (Eb4), (e) Tae (F4), (f) Joong (Ab4), (g) Im (Bb4), and (h) Nam (C5) Fig. 11 demonstrates spectral peaks of the synthesized and original single-notes at Hwangjong position and there is no doubt that they are similar in the shape of their spectra. As shown in Fig. 11, we can obtain the synthesized single-notes with a high fundamental frequency whose spectra are similar to the originals (Tae (F4), Joong (Ab4), Im (Bb4) and Nam (C5)). However, there are some differences for notes with a relatively low fundamental frequency (Joong (Ab3), Im (Bb3), Nam (C4) and Hwang (Eb4)) at several frequencies. They have slightly larger spectral magnitudes than the corresponding original
7 single-notes at below 10 khz and their spectral amplitudes are larger at above 10 khz. These results can be occurred due to the following two reasons: Notes with low fundamental frequencies have many resonances, and thus narrow intervals between resonant frequencies can be measured. In this paper, we decide resonances are efficient when the interval of two adjacent resonances is greater than the fundamental frequency and the amplitude of the current resonance is bigger than the amplitude of adjacent resonances. Consequently, more resonances can be missed at high frequency compared to notes having high fundamental frequencies, and these missing resonances still remain in the residual signal. This results in low-quality synthesized sound. However, this can be solved by adjusting a control parameter to obtain resonances from the original notes. To generate noise components, the sinusoids are subtracted from the original sound in frequency domain. This results in a residual on which the stochastic approximation is performed. We assumed that the residual is a stochastic signal, and consequently it is not necessary to keep exact spectral shape information. This paper presents the linesegment approximation to produce the residual. The line-segment approximation is carried out by finding local maxima at every 100 samples, thus giving equally spaced points in the spectrum that are connected by straight lines to create the spectral envelope. If, however, there are many spectral peaks in the residual, the approximation allows the residual to have high amplitudes for valleys in waveform. To conquer this problem, another type of approximation should be considered. Sound demo samples are available at VI. DISCUSSIONS Spectrum expresses the energy information of input signal in frequency domain, and thus it is influenced by the number of samples in the input signal. As the number of input samples is increased, cepstrum gives the average value of total energy rather than the change over time. Thus, the energy nearby peak (rather than peak) becomes larger because cepstrum is originated in spectrum. Whit this reason, 10 20ms regions of each note are used as input samples to extract exact formants. In this paper, we used sustain region to extract synthesis parameters such as resonant frequencies, magnitudes and bandwidth. However, it is necessary to consider about the usage of attack region as well as the change over time to generate more realistic haegeum sounds. Consequently, we assumed that the input signal is a frame of STFT analysis. Then, we should improve the proposed model into STFT based model which describes the change over time. In spectral modeling, many synthesis parameters are required to synthesize the sounds that are similar to original sounds. However, formant synthesis using spectral envelope can describe the harmonic/inharmonic components, less parameters than additive synthesis in spectral modeling are used, though. Using FFT to extract envelope yields a dense and fine curve but more data are needed. In the case of LPC, the envelope is so smooth that it is very hard to express the characteristics of harmonics of haegeum. Consequently, the higher order of LPC is required to obtain a fine envelope. In cepstral analysis, if cepstral coefficients are separated using a window with an appropriate size, it is possible to have a denser envelope than FFT. As a result, cepstral analysis is more suitable for synthesizing haegeum sounds. Noise components might contain some peaks which are considered as harmonics. To eliminate them, we added an additional process to re-find them in the residual. The synthetic results tend to show magnitudes at higher frequencies are bigger than originals. This is caused by addition of noise components, and thus we need to modify the noise model. VII. CONCULSIONS In this paper, we studied a formant synthesis method of haegeum sounds using cepstral envelope for spectral modeling. The parameters required in synthesis process are reduced by using formant synthesis method instead of additive synthesis that is utilized for sinusoids in existing spectral modeling. Formants are extracted from cepstral envelope based on the characteristics of haegeum, and that is exploited in synthesis of sinusoids. To model noise components, synthesized sinusoids are subtracted from the original signal. To improve synthetic results, we added an additional process to find sinusoidal components remained in the residual. A digital resonator by IIT is used for the synthesis of sinusoids and the line-segment approximation is used for the synthesis of noise. The synthesized sounds are generated by adding sinusoids and noise components, and they are very similar to the originals. REFERENCES [1] C. Roads, The Computer Music Tutorial, The MIT press, London, [2] S. Cho and U. Chong, Sound Synthesis of Right-Hand Playing Styles using Improved Physical Modeling of Sanjo Gayageum, Acoust. Soc. Kor., vol. 25, no. 8, pp , [3] X. Serra and J. O. Smith, "Residual Minimization in a Musical Signal Model based on a Deterministic plus Stochastic Decomposition," J. Acoust. Soc. Am., vol. 95, no. 5 2, pp , [4] X. Serra and J. O. Smith, "Spectral Modeling Synthesis: A Sound Analysis/Synthesis System based on a Deterministic plus Stochastic Decomposition," Comput. Music J., vol. 14, no. 4, pp , [5] Spectral Audio Signal Processing, available at Online Book, [6] H. K. Dennis, "Software for a cascade/parallel formant synthesizer," J. Acoust. Soc. Am., vol. 67, no. 3, pp , [7] A. V. Oppenheim, R. W. Schafer, and J. R. Buck, Ronald W. Schafer, Discrete-Time Signal Processing, 2nd ed., Prentice Hall, [8] X. Serra and J. Bonada, Sound Transformations Based on the SMS High Level Attributes, in Proc. Int l Conf. Digital Audio Effects (DAFX98), [9] T. S. Verma, T. H. Y. Meng, Time Scale Modification Using a Sines+Transients+Noise Signal Model, in Proc. Int l Conf. Digital Audio Effects (DAFX98), [10] T. S. Verma and T. H. Y. Meng, An analysis/synthesis tool for transient signals, in Proc. 16th Int l Cong. Acoustics, vol. 1, pp , [11] T. S. Verma, T. H. Y. Meng., Extending Spectral Modeling Synthesis with Transient Modeling Synthesis, Comput. Music J., vol. 24, no. 2, pp , 2000.
8 [12] J. Noh, S. Park and KM. Sung, Acoustic Characteristics of the Haegeum Body, Acoust. Soc. Kor., vol. 26, no. 7, pp , [13] H. Song, Korean Musical Instruments, 1st ed., Youl Hwa Dang, [14] B. Gold and N. Morgan, Speech and Audio Signal Processing, John Wiley & Sons, 1999.
A Parametric Model for Spectral Sound Synthesis of Musical Sounds
A Parametric Model for Spectral Sound Synthesis of Musical Sounds Cornelia Kreutzer University of Limerick ECE Department Limerick, Ireland cornelia.kreutzer@ul.ie Jacqueline Walker University of Limerick
More informationTHE BEATING EQUALIZER AND ITS APPLICATION TO THE SYNTHESIS AND MODIFICATION OF PIANO TONES
J. Rauhala, The beating equalizer and its application to the synthesis and modification of piano tones, in Proceedings of the 1th International Conference on Digital Audio Effects, Bordeaux, France, 27,
More informationQuantification of glottal and voiced speech harmonicsto-noise ratios using cepstral-based estimation
Quantification of glottal and voiced speech harmonicsto-noise ratios using cepstral-based estimation Peter J. Murphy and Olatunji O. Akande, Department of Electronic and Computer Engineering University
More informationE : Lecture 8 Source-Filter Processing. E : Lecture 8 Source-Filter Processing / 21
E85.267: Lecture 8 Source-Filter Processing E85.267: Lecture 8 Source-Filter Processing 21-4-1 1 / 21 Source-filter analysis/synthesis n f Spectral envelope Spectral envelope Analysis Source signal n 1
More informationSound Synthesis Methods
Sound Synthesis Methods Matti Vihola, mvihola@cs.tut.fi 23rd August 2001 1 Objectives The objective of sound synthesis is to create sounds that are Musically interesting Preferably realistic (sounds like
More informationLecture 5: Sinusoidal Modeling
ELEN E4896 MUSIC SIGNAL PROCESSING Lecture 5: Sinusoidal Modeling 1. Sinusoidal Modeling 2. Sinusoidal Analysis 3. Sinusoidal Synthesis & Modification 4. Noise Residual Dan Ellis Dept. Electrical Engineering,
More informationADDITIVE SYNTHESIS BASED ON THE CONTINUOUS WAVELET TRANSFORM: A SINUSOIDAL PLUS TRANSIENT MODEL
ADDITIVE SYNTHESIS BASED ON THE CONTINUOUS WAVELET TRANSFORM: A SINUSOIDAL PLUS TRANSIENT MODEL José R. Beltrán and Fernando Beltrán Department of Electronic Engineering and Communications University of
More informationAdvanced audio analysis. Martin Gasser
Advanced audio analysis Martin Gasser Motivation Which methods are common in MIR research? How can we parameterize audio signals? Interesting dimensions of audio: Spectral/ time/melody structure, high
More informationspeech signal S(n). This involves a transformation of S(n) into another signal or a set of signals
16 3. SPEECH ANALYSIS 3.1 INTRODUCTION TO SPEECH ANALYSIS Many speech processing [22] applications exploits speech production and perception to accomplish speech analysis. By speech analysis we extract
More informationSpeech Synthesis using Mel-Cepstral Coefficient Feature
Speech Synthesis using Mel-Cepstral Coefficient Feature By Lu Wang Senior Thesis in Electrical Engineering University of Illinois at Urbana-Champaign Advisor: Professor Mark Hasegawa-Johnson May 2018 Abstract
More informationFIR/Convolution. Visulalizing the convolution sum. Convolution
FIR/Convolution CMPT 368: Lecture Delay Effects Tamara Smyth, tamaras@cs.sfu.ca School of Computing Science, Simon Fraser University April 2, 27 Since the feedforward coefficient s of the FIR filter are
More informationNon-stationary Analysis/Synthesis using Spectrum Peak Shape Distortion, Phase and Reassignment
Non-stationary Analysis/Synthesis using Spectrum Peak Shape Distortion, Phase Reassignment Geoffroy Peeters, Xavier Rodet Ircam - Centre Georges-Pompidou, Analysis/Synthesis Team, 1, pl. Igor Stravinsky,
More informationRotating Machinery Fault Diagnosis Techniques Envelope and Cepstrum Analyses
Rotating Machinery Fault Diagnosis Techniques Envelope and Cepstrum Analyses Spectra Quest, Inc. 8205 Hermitage Road, Richmond, VA 23228, USA Tel: (804) 261-3300 www.spectraquest.com October 2006 ABSTRACT
More informationADAPTIVE NOISE LEVEL ESTIMATION
Proc. of the 9 th Int. Conference on Digital Audio Effects (DAFx-6), Montreal, Canada, September 18-2, 26 ADAPTIVE NOISE LEVEL ESTIMATION Chunghsin Yeh Analysis/Synthesis team IRCAM/CNRS-STMS, Paris, France
More informationSignal Processing for Speech Applications - Part 2-1. Signal Processing For Speech Applications - Part 2
Signal Processing for Speech Applications - Part 2-1 Signal Processing For Speech Applications - Part 2 May 14, 2013 Signal Processing for Speech Applications - Part 2-2 References Huang et al., Chapter
More informationSINOLA: A New Analysis/Synthesis Method using Spectrum Peak Shape Distortion, Phase and Reassigned Spectrum
SINOLA: A New Analysis/Synthesis Method using Spectrum Peak Shape Distortion, Phase Reassigned Spectrum Geoffroy Peeters, Xavier Rodet Ircam - Centre Georges-Pompidou Analysis/Synthesis Team, 1, pl. Igor
More informationWARPED FILTER DESIGN FOR THE BODY MODELING AND SOUND SYNTHESIS OF STRING INSTRUMENTS
NORDIC ACOUSTICAL MEETING 12-14 JUNE 1996 HELSINKI WARPED FILTER DESIGN FOR THE BODY MODELING AND SOUND SYNTHESIS OF STRING INSTRUMENTS Helsinki University of Technology Laboratory of Acoustics and Audio
More informationTIME DOMAIN ATTACK AND RELEASE MODELING Applied to Spectral Domain Sound Synthesis
TIME DOMAIN ATTACK AND RELEASE MODELING Applied to Spectral Domain Sound Synthesis Cornelia Kreutzer, Jacqueline Walker Department of Electronic and Computer Engineering, University of Limerick, Limerick,
More informationTopic. Spectrogram Chromagram Cesptrogram. Bryan Pardo, 2008, Northwestern University EECS 352: Machine Perception of Music and Audio
Topic Spectrogram Chromagram Cesptrogram Short time Fourier Transform Break signal into windows Calculate DFT of each window The Spectrogram spectrogram(y,1024,512,1024,fs,'yaxis'); A series of short term
More informationLinguistic Phonetics. Spectral Analysis
24.963 Linguistic Phonetics Spectral Analysis 4 4 Frequency (Hz) 1 Reading for next week: Liljencrants & Lindblom 1972. Assignment: Lip-rounding assignment, due 1/15. 2 Spectral analysis techniques There
More informationMel Spectrum Analysis of Speech Recognition using Single Microphone
International Journal of Engineering Research in Electronics and Communication Mel Spectrum Analysis of Speech Recognition using Single Microphone [1] Lakshmi S.A, [2] Cholavendan M [1] PG Scholar, Sree
More informationHungarian Speech Synthesis Using a Phase Exact HNM Approach
Hungarian Speech Synthesis Using a Phase Exact HNM Approach Kornél Kovács 1, András Kocsor 2, and László Tóth 3 Research Group on Artificial Intelligence of the Hungarian Academy of Sciences and University
More informationFFT analysis in practice
FFT analysis in practice Perception & Multimedia Computing Lecture 13 Rebecca Fiebrink Lecturer, Department of Computing Goldsmiths, University of London 1 Last Week Review of complex numbers: rectangular
More informationLaboratory Assignment 4. Fourier Sound Synthesis
Laboratory Assignment 4 Fourier Sound Synthesis PURPOSE This lab investigates how to use a computer to evaluate the Fourier series for periodic signals and to synthesize audio signals from Fourier series
More informationFIR/Convolution. Visulalizing the convolution sum. Frequency-Domain (Fast) Convolution
FIR/Convolution CMPT 468: Delay Effects Tamara Smyth, tamaras@cs.sfu.ca School of Computing Science, Simon Fraser University November 8, 23 Since the feedforward coefficient s of the FIR filter are the
More informationCMPT 468: Delay Effects
CMPT 468: Delay Effects Tamara Smyth, tamaras@cs.sfu.ca School of Computing Science, Simon Fraser University November 8, 2013 1 FIR/Convolution Since the feedforward coefficient s of the FIR filter are
More informationReading: Johnson Ch , Ch.5.5 (today); Liljencrants & Lindblom; Stevens (Tues) reminder: no class on Thursday.
L105/205 Phonetics Scarborough Handout 7 10/18/05 Reading: Johnson Ch.2.3.3-2.3.6, Ch.5.5 (today); Liljencrants & Lindblom; Stevens (Tues) reminder: no class on Thursday Spectral Analysis 1. There are
More informationMUS421/EE367B Applications Lecture 9C: Time Scale Modification (TSM) and Frequency Scaling/Shifting
MUS421/EE367B Applications Lecture 9C: Time Scale Modification (TSM) and Frequency Scaling/Shifting Julius O. Smith III (jos@ccrma.stanford.edu) Center for Computer Research in Music and Acoustics (CCRMA)
More informationy(n)= Aa n u(n)+bu(n) b m sin(2πmt)= b 1 sin(2πt)+b 2 sin(4πt)+b 3 sin(6πt)+ m=1 x(t)= x = 2 ( b b b b
Exam 1 February 3, 006 Each subquestion is worth 10 points. 1. Consider a periodic sawtooth waveform x(t) with period T 0 = 1 sec shown below: (c) x(n)= u(n). In this case, show that the output has the
More informationVIBRATO DETECTING ALGORITHM IN REAL TIME. Minhao Zhang, Xinzhao Liu. University of Rochester Department of Electrical and Computer Engineering
VIBRATO DETECTING ALGORITHM IN REAL TIME Minhao Zhang, Xinzhao Liu University of Rochester Department of Electrical and Computer Engineering ABSTRACT Vibrato is a fundamental expressive attribute in music,
More informationAudio Engineering Society Convention Paper Presented at the 110th Convention 2001 May Amsterdam, The Netherlands
Audio Engineering Society Convention Paper Presented at the th Convention May 5 Amsterdam, The Netherlands This convention paper has been reproduced from the author's advance manuscript, without editing,
More informationStructure of Speech. Physical acoustics Time-domain representation Frequency domain representation Sound shaping
Structure of Speech Physical acoustics Time-domain representation Frequency domain representation Sound shaping Speech acoustics Source-Filter Theory Speech Source characteristics Speech Filter characteristics
More informationSignal Analysis. Peak Detection. Envelope Follower (Amplitude detection) Music 270a: Signal Analysis
Signal Analysis Music 27a: Signal Analysis Tamara Smyth, trsmyth@ucsd.edu Department of Music, University of California, San Diego (UCSD November 23, 215 Some tools we may want to use to automate analysis
More informationPerformance Analysis of MFCC and LPCC Techniques in Automatic Speech Recognition
www.ijecs.in International Journal Of Engineering And Computer Science ISSN:2319-7242 Volume - 3 Issue - 8 August, 2014 Page No. 7727-7732 Performance Analysis of MFCC and LPCC Techniques in Automatic
More informationPROBLEM SET 6. Note: This version is preliminary in that it does not yet have instructions for uploading the MATLAB problems.
PROBLEM SET 6 Issued: 2/32/19 Due: 3/1/19 Reading: During the past week we discussed change of discrete-time sampling rate, introducing the techniques of decimation and interpolation, which is covered
More informationI-Hao Hsiao, Chun-Tang Chao*, and Chi-Jo Wang (2016). A HHT-Based Music Synthesizer. Intelligent Technologies and Engineering Systems, Lecture Notes
I-Hao Hsiao, Chun-Tang Chao*, and Chi-Jo Wang (2016). A HHT-Based Music Synthesizer. Intelligent Technologies and Engineering Systems, Lecture Notes in Electrical Engineering (LNEE), Vol.345, pp.523-528.
More informationINFLUENCE OF FREQUENCY DISTRIBUTION ON INTENSITY FLUCTUATIONS OF NOISE
INFLUENCE OF FREQUENCY DISTRIBUTION ON INTENSITY FLUCTUATIONS OF NOISE Pierre HANNA SCRIME - LaBRI Université de Bordeaux 1 F-33405 Talence Cedex, France hanna@labriu-bordeauxfr Myriam DESAINTE-CATHERINE
More informationSpeech Synthesis; Pitch Detection and Vocoders
Speech Synthesis; Pitch Detection and Vocoders Tai-Shih Chi ( 冀泰石 ) Department of Communication Engineering National Chiao Tung University May. 29, 2008 Speech Synthesis Basic components of the text-to-speech
More informationVOICE QUALITY SYNTHESIS WITH THE BANDWIDTH ENHANCED SINUSOIDAL MODEL
VOICE QUALITY SYNTHESIS WITH THE BANDWIDTH ENHANCED SINUSOIDAL MODEL Narsimh Kamath Vishweshwara Rao Preeti Rao NIT Karnataka EE Dept, IIT-Bombay EE Dept, IIT-Bombay narsimh@gmail.com vishu@ee.iitb.ac.in
More informationREAL-TIME BROADBAND NOISE REDUCTION
REAL-TIME BROADBAND NOISE REDUCTION Robert Hoeldrich and Markus Lorber Institute of Electronic Music Graz Jakoministrasse 3-5, A-8010 Graz, Austria email: robert.hoeldrich@mhsg.ac.at Abstract A real-time
More informationSOUND SOURCE RECOGNITION AND MODELING
SOUND SOURCE RECOGNITION AND MODELING CASA seminar, summer 2000 Antti Eronen antti.eronen@tut.fi Contents: Basics of human sound source recognition Timbre Voice recognition Recognition of environmental
More informationME scope Application Note 01 The FFT, Leakage, and Windowing
INTRODUCTION ME scope Application Note 01 The FFT, Leakage, and Windowing NOTE: The steps in this Application Note can be duplicated using any Package that includes the VES-3600 Advanced Signal Processing
More informationA Comparative Study of Formant Frequencies Estimation Techniques
A Comparative Study of Formant Frequencies Estimation Techniques DORRA GARGOURI, Med ALI KAMMOUN and AHMED BEN HAMIDA Unité de traitement de l information et électronique médicale, ENIS University of Sfax
More informationCepstrum alanysis of speech signals
Cepstrum alanysis of speech signals ELEC-E5520 Speech and language processing methods Spring 2016 Mikko Kurimo 1 /48 Contents Literature and other material Idea and history of cepstrum Cepstrum and LP
More informationAcoustics, signals & systems for audiology. Week 4. Signals through Systems
Acoustics, signals & systems for audiology Week 4 Signals through Systems Crucial ideas Any signal can be constructed as a sum of sine waves In a linear time-invariant (LTI) system, the response to a sinusoid
More informationInternational Journal of Modern Trends in Engineering and Research e-issn No.: , Date: 2-4 July, 2015
International Journal of Modern Trends in Engineering and Research www.ijmter.com e-issn No.:2349-9745, Date: 2-4 July, 2015 Analysis of Speech Signal Using Graphic User Interface Solly Joy 1, Savitha
More informationIdentification of Nonstationary Audio Signals Using the FFT, with Application to Analysis-based Synthesis of Sound
Identification of Nonstationary Audio Signals Using the FFT, with Application to Analysis-based Synthesis of Sound Paul Masri, Prof. Andrew Bateman Digital Music Research Group, University of Bristol 1.4
More informationSINUSOIDAL MODELING. EE6641 Analysis and Synthesis of Audio Signals. Yi-Wen Liu Nov 3, 2015
1 SINUSOIDAL MODELING EE6641 Analysis and Synthesis of Audio Signals Yi-Wen Liu Nov 3, 2015 2 Last time: Spectral Estimation Resolution Scenario: multiple peaks in the spectrum Choice of window type and
More informationSignal Characterization in terms of Sinusoidal and Non-Sinusoidal Components
Signal Characterization in terms of Sinusoidal and Non-Sinusoidal Components Geoffroy Peeters, avier Rodet To cite this version: Geoffroy Peeters, avier Rodet. Signal Characterization in terms of Sinusoidal
More information4.5 Fractional Delay Operations with Allpass Filters
158 Discrete-Time Modeling of Acoustic Tubes Using Fractional Delay Filters 4.5 Fractional Delay Operations with Allpass Filters The previous sections of this chapter have concentrated on the FIR implementation
More informationSPEECH ANALYSIS-SYNTHESIS FOR SPEAKER CHARACTERISTIC MODIFICATION
M.Tech. Credit Seminar Report, Electronic Systems Group, EE Dept, IIT Bombay, submitted November 04 SPEECH ANALYSIS-SYNTHESIS FOR SPEAKER CHARACTERISTIC MODIFICATION G. Gidda Reddy (Roll no. 04307046)
More informationSound, acoustics Slides based on: Rossing, The science of sound, 1990.
Sound, acoustics Slides based on: Rossing, The science of sound, 1990. Acoustics 1 1 Introduction Acoustics 2! The word acoustics refers to the science of sound and is a subcategory of physics! Room acoustics
More informationSGN Audio and Speech Processing
Introduction 1 Course goals Introduction 2 SGN 14006 Audio and Speech Processing Lectures, Fall 2014 Anssi Klapuri Tampere University of Technology! Learn basics of audio signal processing Basic operations
More informationPR No. 119 DIGITAL SIGNAL PROCESSING XVIII. Academic Research Staff. Prof. Alan V. Oppenheim Prof. James H. McClellan.
XVIII. DIGITAL SIGNAL PROCESSING Academic Research Staff Prof. Alan V. Oppenheim Prof. James H. McClellan Graduate Students Bir Bhanu Gary E. Kopec Thomas F. Quatieri, Jr. Patrick W. Bosshart Jae S. Lim
More informationLecture 6: Nonspeech and Music
EE E682: Speech & Audio Processing & Recognition Lecture 6: Nonspeech and Music 1 Music & nonspeech Dan Ellis Michael Mandel 2 Environmental Sounds Columbia
More information2.1 BASIC CONCEPTS Basic Operations on Signals Time Shifting. Figure 2.2 Time shifting of a signal. Time Reversal.
1 2.1 BASIC CONCEPTS 2.1.1 Basic Operations on Signals Time Shifting. Figure 2.2 Time shifting of a signal. Time Reversal. 2 Time Scaling. Figure 2.4 Time scaling of a signal. 2.1.2 Classification of Signals
More informationINTRODUCTION TO COMPUTER MUSIC SAMPLING SYNTHESIS AND FILTERS. Professor of Computer Science, Art, and Music
INTRODUCTION TO COMPUTER MUSIC SAMPLING SYNTHESIS AND FILTERS Roger B. Dannenberg Professor of Computer Science, Art, and Music Copyright 2002-2013 by Roger B. Dannenberg 1 SAMPLING SYNTHESIS Synthesis
More informationResonator Factoring. Julius Smith and Nelson Lee
Resonator Factoring Julius Smith and Nelson Lee RealSimple Project Center for Computer Research in Music and Acoustics (CCRMA) Department of Music, Stanford University Stanford, California 9435 March 13,
More informationSynthesis Techniques. Juan P Bello
Synthesis Techniques Juan P Bello Synthesis It implies the artificial construction of a complex body by combining its elements. Complex body: acoustic signal (sound) Elements: parameters and/or basic signals
More informationMusical Acoustics, C. Bertulani. Musical Acoustics. Lecture 13 Timbre / Tone quality I
1 Musical Acoustics Lecture 13 Timbre / Tone quality I Waves: review 2 distance x (m) At a given time t: y = A sin(2πx/λ) A -A time t (s) At a given position x: y = A sin(2πt/t) Perfect Tuning Fork: Pure
More informationL19: Prosodic modification of speech
L19: Prosodic modification of speech Time-domain pitch synchronous overlap add (TD-PSOLA) Linear-prediction PSOLA Frequency-domain PSOLA Sinusoidal models Harmonic + noise models STRAIGHT This lecture
More informationECE438 - Laboratory 7a: Digital Filter Design (Week 1) By Prof. Charles Bouman and Prof. Mireille Boutin Fall 2015
Purdue University: ECE438 - Digital Signal Processing with Applications 1 ECE438 - Laboratory 7a: Digital Filter Design (Week 1) By Prof. Charles Bouman and Prof. Mireille Boutin Fall 2015 1 Introduction
More informationFundamentals of Time- and Frequency-Domain Analysis of Signal-Averaged Electrocardiograms R. Martin Arthur, PhD
CORONARY ARTERY DISEASE, 2(1):13-17, 1991 1 Fundamentals of Time- and Frequency-Domain Analysis of Signal-Averaged Electrocardiograms R. Martin Arthur, PhD Keywords digital filters, Fourier transform,
More informationEnhanced Waveform Interpolative Coding at 4 kbps
Enhanced Waveform Interpolative Coding at 4 kbps Oded Gottesman, and Allen Gersho Signal Compression Lab. University of California, Santa Barbara E-mail: [oded, gersho]@scl.ece.ucsb.edu Signal Compression
More informationCOMP 546, Winter 2017 lecture 20 - sound 2
Today we will examine two types of sounds that are of great interest: music and speech. We will see how a frequency domain analysis is fundamental to both. Musical sounds Let s begin by briefly considering
More informationAdvanced Audiovisual Processing Expected Background
Advanced Audiovisual Processing Expected Background As an advanced module, we will not cover introductory topics in lecture. You are expected to already be proficient with all of the following topics,
More informationOverview of Code Excited Linear Predictive Coder
Overview of Code Excited Linear Predictive Coder Minal Mulye 1, Sonal Jagtap 2 1 PG Student, 2 Assistant Professor, Department of E&TC, Smt. Kashibai Navale College of Engg, Pune, India Abstract Advances
More informationAudio Signal Compression using DCT and LPC Techniques
Audio Signal Compression using DCT and LPC Techniques P. Sandhya Rani#1, D.Nanaji#2, V.Ramesh#3,K.V.S. Kiran#4 #Student, Department of ECE, Lendi Institute Of Engineering And Technology, Vizianagaram,
More informationAdaptive noise level estimation
Adaptive noise level estimation Chunghsin Yeh, Axel Roebel To cite this version: Chunghsin Yeh, Axel Roebel. Adaptive noise level estimation. Workshop on Computer Music and Audio Technology (WOCMAT 6),
More informationConverting Speaking Voice into Singing Voice
Converting Speaking Voice into Singing Voice 1 st place of the Synthesis of Singing Challenge 2007: Vocal Conversion from Speaking to Singing Voice using STRAIGHT by Takeshi Saitou et al. 1 STRAIGHT Speech
More informationBiomedical Signals. Signals and Images in Medicine Dr Nabeel Anwar
Biomedical Signals Signals and Images in Medicine Dr Nabeel Anwar Noise Removal: Time Domain Techniques 1. Synchronized Averaging (covered in lecture 1) 2. Moving Average Filters (today s topic) 3. Derivative
More informationUniversity of Washington Department of Electrical Engineering Computer Speech Processing EE516 Winter 2005
University of Washington Department of Electrical Engineering Computer Speech Processing EE516 Winter 2005 Lecture 5 Slides Jan 26 th, 2005 Outline of Today s Lecture Announcements Filter-bank analysis
More informationChapter 5 Window Functions. periodic with a period of N (number of samples). This is observed in table (3.1).
Chapter 5 Window Functions 5.1 Introduction As discussed in section (3.7.5), the DTFS assumes that the input waveform is periodic with a period of N (number of samples). This is observed in table (3.1).
More informationDesign of a Sharp Linear-Phase FIR Filter Using the α-scaled Sampling Kernel
Proceedings of the 6th WSEAS International Conference on SIGNAL PROCESSING, Dallas, Texas, USA, March 22-24, 2007 129 Design of a Sharp Linear-Phase FIR Filter Using the -scaled Sampling Kernel K.J. Kim,
More informationSpeech Enhancement Using Spectral Flatness Measure Based Spectral Subtraction
IOSR Journal of VLSI and Signal Processing (IOSR-JVSP) Volume 7, Issue, Ver. I (Mar. - Apr. 7), PP 4-46 e-issn: 9 4, p-issn No. : 9 497 www.iosrjournals.org Speech Enhancement Using Spectral Flatness Measure
More informationAdaptive Filters Application of Linear Prediction
Adaptive Filters Application of Linear Prediction Gerhard Schmidt Christian-Albrechts-Universität zu Kiel Faculty of Engineering Electrical Engineering and Information Technology Digital Signal Processing
More informationMeasurement System for Acoustic Absorption Using the Cepstrum Technique. Abstract. 1. Introduction
The 00 International Congress and Exposition on Noise Control Engineering Dearborn, MI, USA. August 9-, 00 Measurement System for Acoustic Absorption Using the Cepstrum Technique E.R. Green Roush Industries
More informationDifferent Approaches of Spectral Subtraction Method for Speech Enhancement
ISSN 2249 5460 Available online at www.internationalejournals.com International ejournals International Journal of Mathematical Sciences, Technology and Humanities 95 (2013 1056 1062 Different Approaches
More informationB.Tech III Year II Semester (R13) Regular & Supplementary Examinations May/June 2017 DIGITAL SIGNAL PROCESSING (Common to ECE and EIE)
Code: 13A04602 R13 B.Tech III Year II Semester (R13) Regular & Supplementary Examinations May/June 2017 (Common to ECE and EIE) PART A (Compulsory Question) 1 Answer the following: (10 X 02 = 20 Marks)
More information8.3 Basic Parameters for Audio
8.3 Basic Parameters for Audio Analysis Physical audio signal: simple one-dimensional amplitude = loudness frequency = pitch Psycho-acoustic features: complex A real-life tone arises from a complex superposition
More informationSound Modeling from the Analysis of Real Sounds
Sound Modeling from the Analysis of Real Sounds S lvi Ystad Philippe Guillemain Richard Kronland-Martinet CNRS, Laboratoire de Mécanique et d'acoustique 31, Chemin Joseph Aiguier, 13402 Marseille cedex
More informationFinal Exam Practice Questions for Music 421, with Solutions
Final Exam Practice Questions for Music 4, with Solutions Elementary Fourier Relationships. For the window w = [/,,/ ], what is (a) the dc magnitude of the window transform? + (b) the magnitude at half
More informationSince the advent of the sine wave oscillator
Advanced Distortion Analysis Methods Discover modern test equipment that has the memory and post-processing capability to analyze complex signals and ascertain real-world performance. By Dan Foley European
More informationDigital Signal Processing
Digital Signal Processing Fourth Edition John G. Proakis Department of Electrical and Computer Engineering Northeastern University Boston, Massachusetts Dimitris G. Manolakis MIT Lincoln Laboratory Lexington,
More informationSignal processing preliminaries
Signal processing preliminaries ISMIR Graduate School, October 4th-9th, 2004 Contents: Digital audio signals Fourier transform Spectrum estimation Filters Signal Proc. 2 1 Digital signals Advantages of
More informationFinal Exam Study Guide: Introduction to Computer Music Course Staff April 24, 2015
Final Exam Study Guide: 15-322 Introduction to Computer Music Course Staff April 24, 2015 This document is intended to help you identify and master the main concepts of 15-322, which is also what we intend
More informationPreeti Rao 2 nd CompMusicWorkshop, Istanbul 2012
Preeti Rao 2 nd CompMusicWorkshop, Istanbul 2012 o Music signal characteristics o Perceptual attributes and acoustic properties o Signal representations for pitch detection o STFT o Sinusoidal model o
More informationTHE CITADEL THE MILITARY COLLEGE OF SOUTH CAROLINA. Department of Electrical and Computer Engineering. ELEC 423 Digital Signal Processing
THE CITADEL THE MILITARY COLLEGE OF SOUTH CAROLINA Department of Electrical and Computer Engineering ELEC 423 Digital Signal Processing Project 2 Due date: November 12 th, 2013 I) Introduction In ELEC
More information19 th INTERNATIONAL CONGRESS ON ACOUSTICS MADRID, 2-7 SEPTEMBER 2007
19 th INTERNATIONAL CONGRESS ON ACOUSTICS MADRID, 2-7 SEPTEMBER 2007 Acoustic Radiation Pattern of the Sanjo Gayageum: A Korean traditional plucked string instrument PACS: 43.75.-z Jung Uk Noh; Hyun-Woo
More informationANALYSIS OF PIANO TONES USING AN INHARMONIC INVERSE COMB FILTER
Proc. of the 11 th Int. Conference on Digital Audio Effects (DAFx-8), Espoo, Finland, September 1-4, 28 ANALYSIS OF PIANO TONES USING AN INHARMONIC INVERSE COMB FILTER Heidi-Maria Lehtonen Department of
More informationTRANSFORMS / WAVELETS
RANSFORMS / WAVELES ransform Analysis Signal processing using a transform analysis for calculations is a technique used to simplify or accelerate problem solution. For example, instead of dividing two
More informationKhlui-Phiang-Aw Sound Synthesis Using A Warped FIR Filter
Khlui-Phiang-Aw Sound Synthesis Using A Warped FIR Filter Korakoch Saengrattanakul Faculty of Engineering, Khon Kaen University Khon Kaen-40002, Thailand. ORCID: 0000-0001-8620-8782 Kittipitch Meesawat*
More informationVocoder (LPC) Analysis by Variation of Input Parameters and Signals
ISCA Journal of Engineering Sciences ISCA J. Engineering Sci. Vocoder (LPC) Analysis by Variation of Input Parameters and Signals Abstract Gupta Rajani, Mehta Alok K. and Tiwari Vebhav Truba College of
More informationSound pressure level calculation methodology investigation of corona noise in AC substations
International Conference on Advanced Electronic Science and Technology (AEST 06) Sound pressure level calculation methodology investigation of corona noise in AC substations,a Xiaowen Wu, Nianguang Zhou,
More informationComparison of Multirate two-channel Quadrature Mirror Filter Bank with FIR Filters Based Multiband Dynamic Range Control for audio
IOSR Journal of Electronics and Communication Engineering (IOSR-JECE) e-issn: 2278-2834,p- ISSN: 2278-8735.Volume 9, Issue 3, Ver. IV (May - Jun. 2014), PP 19-24 Comparison of Multirate two-channel Quadrature
More informationSubtractive Synthesis. Describing a Filter. Filters. CMPT 468: Subtractive Synthesis
Subtractive Synthesis CMPT 468: Subtractive Synthesis Tamara Smyth, tamaras@cs.sfu.ca School of Computing Science, Simon Fraser University November, 23 Additive synthesis involves building the sound by
More informationAudio Restoration Based on DSP Tools
Audio Restoration Based on DSP Tools EECS 451 Final Project Report Nan Wu School of Electrical Engineering and Computer Science University of Michigan Ann Arbor, MI, United States wunan@umich.edu Abstract
More informationSynchronous Overlap and Add of Spectra for Enhancement of Excitation in Artificial Bandwidth Extension of Speech
INTERSPEECH 5 Synchronous Overlap and Add of Spectra for Enhancement of Excitation in Artificial Bandwidth Extension of Speech M. A. Tuğtekin Turan and Engin Erzin Multimedia, Vision and Graphics Laboratory,
More informationEE482: Digital Signal Processing Applications
Professor Brendan Morris, SEB 3216, brendan.morris@unlv.edu EE482: Digital Signal Processing Applications Spring 2014 TTh 14:30-15:45 CBC C222 Lecture 12 Speech Signal Processing 14/03/25 http://www.ee.unlv.edu/~b1morris/ee482/
More informationPitch Period of Speech Signals Preface, Determination and Transformation
Pitch Period of Speech Signals Preface, Determination and Transformation Mohammad Hossein Saeidinezhad 1, Bahareh Karamsichani 2, Ehsan Movahedi 3 1 Islamic Azad university, Najafabad Branch, Saidinezhad@yahoo.com
More information