Linear predictive coding (LPC) models human speech production as a vocal tract represented by a tube of varying diameter. LPC is a digital encoding method that predicts signal values using a linear function of past values. The speech production process can be modeled as excitation of the vocal tract filter by either a periodic source for voiced sounds or random noise for unvoiced sounds. Accurately modeling the short-term power spectral envelope plays an important role in speech quality and intelligibility. The Itakura-Saito distance measure is an error criterion used in LPC that better models periodic signals like voiced speech. An LPC vocoder separates speech into parameters at the transmitter and re synthesizes speech from the parameters at the receiver.
Analysis of PEAQ Model using Wavelet Decomposition Techniquesidescitation
Digital broadcasting, internet audio and music database make use of audio
compression and coding techniques to reduce high quality audio signal without impairing its
perceptual quality. Audio signal compression is the lossy compression
technique, It
converts original converting audio signal into compressed bitstream. The compressed audio
bitstream is decoded at the decoder to produce a close approximation of the original signal.
For the purpose of improving the coding this work attempts to verify the perceptual
evaluation of audio quality (PEAQ) model in BS.1387 using wavelet decomposition
techniques. Finally the comparison of masking threshold for sub-bands using Wavelet
techniques and Fast Fourier transform (FFT) will be done
Tutorial on neural vocoders at the 2021 Speech Processing Courses in Crete, "Inclusive Neural Speech Synthesis."
Presenters: Xin Wang and Junichi Yamagishi, National Institute of Informatics, Japan
PROGRAMMA ATTIVITA’ DIDATTICA A.A. 2016/17
DOTTORATO DI RICERCA IN INGEGNERIA STRUTTURALE E GEOTECNICA
____________________________________________________________
STOCHASTIC DYNAMICS AND MONTE CARLO SIMULATION IN EARTHQUAKE ENGINEERING APPLICATIONS
Lecture Series by
Agathoklis Giaralis, Ph.D., M.ASCE., P.E. City, University of London
Visiting Professor Sapienza University of Rome
Spectral Analysis of Sample Rate ConverterCSCJournals
The aim of digital sample rate conversion is to bring a digital audio signal from one sample frequency to another. The distortion of the audio signal introduced by the sample rate converter should be as low as possible. The generation of the output samples from the input samples may be performed by the application of various methods. In this paper, a new technique of digital sample-rate converter is proposed. We perform the spectral analysis of proposed digital sample rate converter.
We present a causal speech enhancement model working on the
raw waveform that runs in real-time on a laptop CPU. The proposed model is based on an encoder-decoder architecture with
skip-connections. It is optimized on both time and frequency
domains, using multiple loss functions. Empirical evidence
shows that it is capable of removing various kinds of background noise including stationary and non-stationary noises,
as well as room reverb. Additionally, we suggest a set of
data augmentation techniques applied directly on the raw waveform which further improve model performance and its generalization abilities. We perform evaluations on several standard
benchmarks, both using objective metrics and human judgements. The proposed model matches state-of-the-art performance of both causal and non causal methods while working
directly on the raw waveform.
Index Terms: Speech enhancement, speech denoising, neural
networks, raw waveform
Analysis of PEAQ Model using Wavelet Decomposition Techniquesidescitation
Digital broadcasting, internet audio and music database make use of audio
compression and coding techniques to reduce high quality audio signal without impairing its
perceptual quality. Audio signal compression is the lossy compression
technique, It
converts original converting audio signal into compressed bitstream. The compressed audio
bitstream is decoded at the decoder to produce a close approximation of the original signal.
For the purpose of improving the coding this work attempts to verify the perceptual
evaluation of audio quality (PEAQ) model in BS.1387 using wavelet decomposition
techniques. Finally the comparison of masking threshold for sub-bands using Wavelet
techniques and Fast Fourier transform (FFT) will be done
Tutorial on neural vocoders at the 2021 Speech Processing Courses in Crete, "Inclusive Neural Speech Synthesis."
Presenters: Xin Wang and Junichi Yamagishi, National Institute of Informatics, Japan
PROGRAMMA ATTIVITA’ DIDATTICA A.A. 2016/17
DOTTORATO DI RICERCA IN INGEGNERIA STRUTTURALE E GEOTECNICA
____________________________________________________________
STOCHASTIC DYNAMICS AND MONTE CARLO SIMULATION IN EARTHQUAKE ENGINEERING APPLICATIONS
Lecture Series by
Agathoklis Giaralis, Ph.D., M.ASCE., P.E. City, University of London
Visiting Professor Sapienza University of Rome
Spectral Analysis of Sample Rate ConverterCSCJournals
The aim of digital sample rate conversion is to bring a digital audio signal from one sample frequency to another. The distortion of the audio signal introduced by the sample rate converter should be as low as possible. The generation of the output samples from the input samples may be performed by the application of various methods. In this paper, a new technique of digital sample-rate converter is proposed. We perform the spectral analysis of proposed digital sample rate converter.
We present a causal speech enhancement model working on the
raw waveform that runs in real-time on a laptop CPU. The proposed model is based on an encoder-decoder architecture with
skip-connections. It is optimized on both time and frequency
domains, using multiple loss functions. Empirical evidence
shows that it is capable of removing various kinds of background noise including stationary and non-stationary noises,
as well as room reverb. Additionally, we suggest a set of
data augmentation techniques applied directly on the raw waveform which further improve model performance and its generalization abilities. We perform evaluations on several standard
benchmarks, both using objective metrics and human judgements. The proposed model matches state-of-the-art performance of both causal and non causal methods while working
directly on the raw waveform.
Index Terms: Speech enhancement, speech denoising, neural
networks, raw waveform
Computer aided design of communication systems / Simulation Communication Sys...Makan Mohammadi
The report introduces how to use computer simulation in the design of physical layer transmission protocols that are too complex for a purely analytical approach. The goal of the report is to offer the theoretical and practical tools for performing modeling, analysis, and design of the physical level of wireless transmission systems including cellular and personal communication systems, satellite systems and radio relay links. This course is useful for telecommunication systems designers, ICT researchers and experts in the development and design of telecommunication physical layer protocols.
Gene's law, Common gate, kernel Principal Component Analysis, ASIC Physical Design Post-Layout Verification, TSMC180nm, 0.13um IBM CMOS technology, Cadence Virtuoso, FPAA, in Spanish, Bruun E,
The following resources come from the 2009/10 BEng (Hons) in Digital Communications & Electronics (course number 2ELE0064) from the University of Hertfordshire. All the mini projects are designed as level two modules of the undergraduate programmes.
The objective of this module is to have built communication links using existing AM modulation, PSK modulation and demodulation blocks, constructed AM modulators and constructed PSK modulators using operational function blocks based on their mathematical expressions, and conducted simulations of the links and modulators, all in Simulink®.
Computer aided design of communication systems / Simulation Communication Sys...Makan Mohammadi
The report introduces how to use computer simulation in the design of physical layer transmission protocols that are too complex for a purely analytical approach. The goal of the report is to offer the theoretical and practical tools for performing modeling, analysis, and design of the physical level of wireless transmission systems including cellular and personal communication systems, satellite systems and radio relay links. This course is useful for telecommunication systems designers, ICT researchers and experts in the development and design of telecommunication physical layer protocols.
Gene's law, Common gate, kernel Principal Component Analysis, ASIC Physical Design Post-Layout Verification, TSMC180nm, 0.13um IBM CMOS technology, Cadence Virtuoso, FPAA, in Spanish, Bruun E,
The following resources come from the 2009/10 BEng (Hons) in Digital Communications & Electronics (course number 2ELE0064) from the University of Hertfordshire. All the mini projects are designed as level two modules of the undergraduate programmes.
The objective of this module is to have built communication links using existing AM modulation, PSK modulation and demodulation blocks, constructed AM modulators and constructed PSK modulators using operational function blocks based on their mathematical expressions, and conducted simulations of the links and modulators, all in Simulink®.
1.Wireless Communication System_Wireless communication is a broad term that i...JeyaPerumal1
Wireless communication involves the transmission of information over a distance without the help of wires, cables or any other forms of electrical conductors.
Wireless communication is a broad term that incorporates all procedures and forms of connecting and communicating between two or more devices using a wireless signal through wireless communication technologies and devices.
Features of Wireless Communication
The evolution of wireless technology has brought many advancements with its effective features.
The transmitted distance can be anywhere between a few meters (for example, a television's remote control) and thousands of kilometers (for example, radio communication).
Wireless communication can be used for cellular telephony, wireless access to the internet, wireless home networking, and so on.
This 7-second Brain Wave Ritual Attracts Money To You.!nirahealhty
Discover the power of a simple 7-second brain wave ritual that can attract wealth and abundance into your life. By tapping into specific brain frequencies, this technique helps you manifest financial success effortlessly. Ready to transform your financial future? Try this powerful ritual and start attracting money today!
ER(Entity Relationship) Diagram for online shopping - TAEHimani415946
https://bit.ly/3KACoyV
The ER diagram for the project is the foundation for the building of the database of the project. The properties, datatypes, and attributes are defined by the ER diagram.
Multi-cluster Kubernetes Networking- Patterns, Projects and GuidelinesSanjeev Rampal
Talk presented at Kubernetes Community Day, New York, May 2024.
Technical summary of Multi-Cluster Kubernetes Networking architectures with focus on 4 key topics.
1) Key patterns for Multi-cluster architectures
2) Architectural comparison of several OSS/ CNCF projects to address these patterns
3) Evolution trends for the APIs of these projects
4) Some design recommendations & guidelines for adopting/ deploying these solutions.
2. PREDICTIVE MODELING OF SPEECH
Linear Predictive Coding (LPC):
is defined as a digital method for encoding an analog signal in
which a particular value is predicted by a linear function of
the past values of the signal
- First proposed method for encoding Human Speech US DoD
- Human speech is produced in the vocal tract which can be
approximated as a variable diameter tube.
- The linear predictive coding (LPC) model is based on a
mathematical approximation of the vocal tract represented
by a tube of a varying diameter
4. Ø The model assumes that the sound-generating mechanism
is linearly separable from the intelligence-modulating
vocal-tract filter.
Ø The precise form of the excitation depends on whether the
Speech is voiced or unvoiced.
• Voiced speech sound (such as /i/ in eve) is generated
from a quasi-periodic excitation of vocal-tract.
• Unvoiced speech sound (such as /f/ in fish) is generated
from random sounds produced by turbulent airflow
through a constriction along the vocal tract.
SPEECH PRODUCTION
5. SPEECH PRODUCTION
Vocal-tract filter is represented by the all-pole transfer function,
å
=
-
+
= M
k
k
k z
a
G
z
H
1
1
)
(
where
G: Gain factor
ak: Real-value coefficient
The form of excitation applied to this filter is changed by
switching between the voiced and unvoiced sounds.
6. SPEECH PRODUCTION
ü Accurate modeling the short-term power spectral
envelope plays an important role in the quality and
intelligibility of the reconstructed speech.
ü At low bit-rate speech coding, an all-pole filter is
adopted to model the spectral information in LPC
(linear predictive coding)-based coders.
ü By minimizing the MSE between the actual speech
samples and the predicted ones, an optimal set of
weights for an all-pole (synthesis) digital filter can
be determined
7. Assuming an “all-pole” model, irrespective of speech signal
envelope we are trying to estimate:
• If sound is unvoiced, usual LPC method gives “good
estimate” of filter coefficients ak
• If sound is voiced, usual LPC method gives a “biased
estimate” of ak with the estimate worsening as the
pitch increases (error criterion being MSE)
Example: This example illustrates the limitations of standard
all-pole modeling of periodic waveforms. Specifications are:
• All-pole filter order, M = 12
• Input Signal: periodic pulse sequence with N=32
SPEECH PRODUCTION
8. The solid line is the original 12-pole envelope, which goes through all the points.
The dashed line is the 12-pole linear predictive model for N=30 spectral lines
SPEECH PRODUCTION
9. Itakura-Saito distance measure:
Let u(n) be a real-valued, stationary stochastic process, it’s
Fourier Transform Uk is:
( ) 1
-
N
...
0,1,2,
k
1
0
=
= å
-
=
-
N
k
jn
n
k
k
e
u
U w
El-jaroudi and Makhoul used “Itakura-Saito distance measure”
as “error criterion” to overcome this problem to develop a
new model called “discrete all-pole model”.
SPEECH PRODUCTION
10. Let vector a denote a set of spectral parameters as:
T
M
a
a
a ]
,.....,
,
[
a 2
1
=
Note: See p.no.189-191 in textbook for proof
The auto-correlation function of u(n) for lag m is:
( )
( ) 1
-
N
...
0,1,2,
k
)
(
where
1
-
N
...
0,1,2,
k
1
)
(
1
0
1
0
=
=
=
=
å
å
-
=
-
-
=
N
k
jn
k
N
k
jn
k
k
k
e
m
r
S
e
S
N
m
r
w
w
SPEECH PRODUCTION
11. The time-reversed impulse response h(-i) of the discrete
frequency-sampled all-pole filter can be obtained from the
relation of predictor coefficients to the auto-correlation as:
å
=
-
=
-
M
k
k k
i
r
a
i
h
0
)
(
)
(
ˆ
å
-
=
÷
÷
ø
ö
ç
ç
è
æ
-
÷
÷
ø
ö
ç
ç
è
æ
-
=
1
0
2
2
1
)
(
ln
)
(
)
(
N
k k
k
k
k
IS
a
S
U
a
S
U
a
D
Itakura-Saito distance measure is given by:
SPEECH PRODUCTION
12. LPC Vocoder
LPC system for digital transmission and reception of speech
signals over a communication channel
Reproduction
of Speech Signal
From
Channel
Output
Speech
Signal
To
Channel
input
Window
LPC
Analyzer
Coder
Pitch
Detector
Block diagram of LPC Vocoder : a) Transmitter b) receiver
(b)
Decoder
Speech
Synthesizer
(a)
13. LPC Vocoder
Transmitter:
• Applies window (typically, 10 to 30 ms long) to the input
• Analyzes the input speech block by block
Ø Performs Linear prediction
Ø Pitch Detection
• Finally, following parameters are encoded:
Ø Set of coefficients computed by LPC Analyzer
Ø The Pitch Period
Ø The Gain parameter
Ø The voiced-unvoiced parameters
14. LPC Vocoder
Receiver:
Performs the inverse operations on the channel output:
• Decode incoming parameters
• Synthesize a speech signal from the parameters