International Journal of Engineering Research and Applications (IJERA) is an open access online peer reviewed international journal that publishes research and review articles in the fields of Computer Science, Neural Networks, Electrical Engineering, Software Engineering, Information Technology, Mechanical Engineering, Chemical Engineering, Plastic Engineering, Food Technology, Textile Engineering, Nano Technology & science, Power Electronics, Electronics & Communication Engineering, Computational mathematics, Image processing, Civil Engineering, Structural Engineering, Environmental Engineering, VLSI Testing & Low Power VLSI Design etc.
Audio Morphing for Percussive Sound Generationa3labdsp
The aim of audio morphing algorithms is to combine two or more sounds to create a new sound with intermediate timbre and duration. During the last two decades several efforts have been made to improve morphing algorithms in order to obtain more realistic and perceptually relevant sounds. In this paper we present an automatic audio morphing technique applied to percussive musical instruments. Based on preprocessing of the sound references in frequency domain and linear interpolation in time domain, the presented approach allows one to generate high quality hybrid sounds at a low computational cost. Several results are reported in order to show the effectiveness of the proposed approach in terms of audio quality and acoustic perception of the generated hybrid sounds, taking into consideration different percussive samples. Mean opinion score and multidimensional scaling were used to compare the presented approach with existing state of the art techniques.
Noise reduction is the process of removing noise from a signal. In this project, two audio files are given: (1) speech.au and (2) noisy_speech.au. The first file contains the original speech signal and the second one contains the noisy version of the first signal. The objective of this project is to reduce the noise from the noisy file
Echo and reverberation effects are used extensively in the music industry. Here we will design a digital filter that will create the echo and reverb effect on audio signals.
Audio Morphing for Percussive Sound Generationa3labdsp
The aim of audio morphing algorithms is to combine two or more sounds to create a new sound with intermediate timbre and duration. During the last two decades several efforts have been made to improve morphing algorithms in order to obtain more realistic and perceptually relevant sounds. In this paper we present an automatic audio morphing technique applied to percussive musical instruments. Based on preprocessing of the sound references in frequency domain and linear interpolation in time domain, the presented approach allows one to generate high quality hybrid sounds at a low computational cost. Several results are reported in order to show the effectiveness of the proposed approach in terms of audio quality and acoustic perception of the generated hybrid sounds, taking into consideration different percussive samples. Mean opinion score and multidimensional scaling were used to compare the presented approach with existing state of the art techniques.
Noise reduction is the process of removing noise from a signal. In this project, two audio files are given: (1) speech.au and (2) noisy_speech.au. The first file contains the original speech signal and the second one contains the noisy version of the first signal. The objective of this project is to reduce the noise from the noisy file
Echo and reverberation effects are used extensively in the music industry. Here we will design a digital filter that will create the echo and reverb effect on audio signals.
echo types, how to cancel echo in each type, which is more complex, echo cancellation implementation in matlab
prepared by : OLA MASHAQI ,, SUHAD MALAYSHE
IJRET : International Journal of Research in Engineering and Technology is an international peer reviewed, online journal published by eSAT Publishing House for the enhancement of research in various disciplines of Engineering and Technology. The aim and scope of the journal is to provide an academic medium and an important reference for the advancement and dissemination of research results that support high-level learning, teaching and research in the fields of Engineering and Technology. We bring together Scientists, Academician, Field Engineers, Scholars and Students of related fields of Engineering and Technology.
Digital signal processing through speech, hearing, and PythonMel Chua
Slides from PyCon 2013 tutorial reformatted for self-study. Code at https://github.com/mchua/pycon-sigproc, original description follows: Why do pianos sound different from guitars? How can we visualize how deafness affects a child's speech? These are signal processing questions, traditionally tackled only by upper-level engineering students with MATLAB and differential equations; we're going to do it with algebra and basic Python skills. Based on a signal processing class for audiology graduate students, taught by a deaf musician.
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
Audio Noise Removal – The State of the Artijceronline
International Journal of Computational Engineering Research (IJCER) is dedicated to protecting personal information and will make every reasonable effort to handle collected information appropriately. All information collected, as well as related requests, will be handled as carefully and efficiently as possible in accordance with IJCER standards for integrity and objectivity.
Feature Extraction of Musical Instrument Tones using FFT and Segment AveragingTELKOMNIKA JOURNAL
A feature extraction for musical instrument tones that based on a transform domain approach was
proposed in this paper. The aim of the proposed feature extraction was to get the lower feature extraction
coefficients. In general, the proposed feature extraction was carried out as follow. Firstly, the input signal
was transformed using FFT (Fast Fourier Transform). Secondly, the left half of the transformed signal was
divided into a number of segments. Finally, the averaging results of that segments, was the feature
extraction of the input signal. Based on the test results, the proposed feature extraction was highly efficient
for the tones, which have many significant local peaks in the Fourier transform domain, because it only
required at least four feature extraction coefficients, in order to represent every tone.
Linear Predictive Coding (LPC) is one of the most powerful speech analysis techniques, and one of the most useful methods for encoding good quality speech at a low bit rate. It provides extremely accurate estimates of speech parameters, and is relatively efficient for computation.
Hybrid Reverberation Algorithm: a Practical Approacha3labdsp
Reverberation is a well known eect that has an important role in our listening experience. Reverberation changes positively the perception of the sound, adding fullness and sense of space. Generally, two approaches are employed for articial reverberation: the desired signal can be obtained by convolving the input signal
with a measured impulse response (IR) or by synthetic techniques based on recursive lter structures. Taking into account the advantages of both approaches, a hybrid articial reverberation algorithm is presented aiming to reproduce the acoustic behaviour of real environment with a low computational load. More in detail, the early reflections are derived from a real impulse response, truncated considering the calculated mixing time, and the reverberation tail is obtained using an IIR lter network. The parameters dening this structure are automatically derived from the analyzed impulse response, using a minimization criteria based on Simultaneous Perturbation Stochastic Approximation (SPSA). The effectiveness of the proposed approach has been proved taking into account a real Italian Theatre impulse response providing comparison with the existing state-of-art techniques in terms of objective and subjective measures.
In this paper, the performances of adaptive noise cancelling system employing Least Mean Square (LMS) algorithm are studied considering both white Gaussian noise (Case 1) and colored noise (Case 2)
situations. Performance is analysed with varying number of iterations, Signal to Noise Ratio (SNR) and tap size with considering Mean Square Error (MSE) as the performance measurement criteria. Results show that the noise reduction is better as well as convergence speed is faster for Case 2 as compared with Case 1. It is also observed that MSE decreases with increasing SNR with relatively faster decrease of MSE in Case 2 as compared with Case 1, and on average MSE increases linearly with increasing number of filter
coefficients for both type of noise situations. All the experiments have been done using computer
simulations implemented on MATLAB platform.
International Journal of Engineering Research and Applications (IJERA) is an open access online peer reviewed international journal that publishes research and review articles in the fields of Computer Science, Neural Networks, Electrical Engineering, Software Engineering, Information Technology, Mechanical Engineering, Chemical Engineering, Plastic Engineering, Food Technology, Textile Engineering, Nano Technology & science, Power Electronics, Electronics & Communication Engineering, Computational mathematics, Image processing, Civil Engineering, Structural Engineering, Environmental Engineering, VLSI Testing & Low Power VLSI Design etc.
echo types, how to cancel echo in each type, which is more complex, echo cancellation implementation in matlab
prepared by : OLA MASHAQI ,, SUHAD MALAYSHE
IJRET : International Journal of Research in Engineering and Technology is an international peer reviewed, online journal published by eSAT Publishing House for the enhancement of research in various disciplines of Engineering and Technology. The aim and scope of the journal is to provide an academic medium and an important reference for the advancement and dissemination of research results that support high-level learning, teaching and research in the fields of Engineering and Technology. We bring together Scientists, Academician, Field Engineers, Scholars and Students of related fields of Engineering and Technology.
Digital signal processing through speech, hearing, and PythonMel Chua
Slides from PyCon 2013 tutorial reformatted for self-study. Code at https://github.com/mchua/pycon-sigproc, original description follows: Why do pianos sound different from guitars? How can we visualize how deafness affects a child's speech? These are signal processing questions, traditionally tackled only by upper-level engineering students with MATLAB and differential equations; we're going to do it with algebra and basic Python skills. Based on a signal processing class for audiology graduate students, taught by a deaf musician.
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
Audio Noise Removal – The State of the Artijceronline
International Journal of Computational Engineering Research (IJCER) is dedicated to protecting personal information and will make every reasonable effort to handle collected information appropriately. All information collected, as well as related requests, will be handled as carefully and efficiently as possible in accordance with IJCER standards for integrity and objectivity.
Feature Extraction of Musical Instrument Tones using FFT and Segment AveragingTELKOMNIKA JOURNAL
A feature extraction for musical instrument tones that based on a transform domain approach was
proposed in this paper. The aim of the proposed feature extraction was to get the lower feature extraction
coefficients. In general, the proposed feature extraction was carried out as follow. Firstly, the input signal
was transformed using FFT (Fast Fourier Transform). Secondly, the left half of the transformed signal was
divided into a number of segments. Finally, the averaging results of that segments, was the feature
extraction of the input signal. Based on the test results, the proposed feature extraction was highly efficient
for the tones, which have many significant local peaks in the Fourier transform domain, because it only
required at least four feature extraction coefficients, in order to represent every tone.
Linear Predictive Coding (LPC) is one of the most powerful speech analysis techniques, and one of the most useful methods for encoding good quality speech at a low bit rate. It provides extremely accurate estimates of speech parameters, and is relatively efficient for computation.
Hybrid Reverberation Algorithm: a Practical Approacha3labdsp
Reverberation is a well known eect that has an important role in our listening experience. Reverberation changes positively the perception of the sound, adding fullness and sense of space. Generally, two approaches are employed for articial reverberation: the desired signal can be obtained by convolving the input signal
with a measured impulse response (IR) or by synthetic techniques based on recursive lter structures. Taking into account the advantages of both approaches, a hybrid articial reverberation algorithm is presented aiming to reproduce the acoustic behaviour of real environment with a low computational load. More in detail, the early reflections are derived from a real impulse response, truncated considering the calculated mixing time, and the reverberation tail is obtained using an IIR lter network. The parameters dening this structure are automatically derived from the analyzed impulse response, using a minimization criteria based on Simultaneous Perturbation Stochastic Approximation (SPSA). The effectiveness of the proposed approach has been proved taking into account a real Italian Theatre impulse response providing comparison with the existing state-of-art techniques in terms of objective and subjective measures.
In this paper, the performances of adaptive noise cancelling system employing Least Mean Square (LMS) algorithm are studied considering both white Gaussian noise (Case 1) and colored noise (Case 2)
situations. Performance is analysed with varying number of iterations, Signal to Noise Ratio (SNR) and tap size with considering Mean Square Error (MSE) as the performance measurement criteria. Results show that the noise reduction is better as well as convergence speed is faster for Case 2 as compared with Case 1. It is also observed that MSE decreases with increasing SNR with relatively faster decrease of MSE in Case 2 as compared with Case 1, and on average MSE increases linearly with increasing number of filter
coefficients for both type of noise situations. All the experiments have been done using computer
simulations implemented on MATLAB platform.
International Journal of Engineering Research and Applications (IJERA) is an open access online peer reviewed international journal that publishes research and review articles in the fields of Computer Science, Neural Networks, Electrical Engineering, Software Engineering, Information Technology, Mechanical Engineering, Chemical Engineering, Plastic Engineering, Food Technology, Textile Engineering, Nano Technology & science, Power Electronics, Electronics & Communication Engineering, Computational mathematics, Image processing, Civil Engineering, Structural Engineering, Environmental Engineering, VLSI Testing & Low Power VLSI Design etc.
International Journal of Engineering Research and Applications (IJERA) is an open access online peer reviewed international journal that publishes research and review articles in the fields of Computer Science, Neural Networks, Electrical Engineering, Software Engineering, Information Technology, Mechanical Engineering, Chemical Engineering, Plastic Engineering, Food Technology, Textile Engineering, Nano Technology & science, Power Electronics, Electronics & Communication Engineering, Computational mathematics, Image processing, Civil Engineering, Structural Engineering, Environmental Engineering, VLSI Testing & Low Power VLSI Design etc.
On 14th July 2014 a memorial symposium to celebrate the life and work of Professor Jean-Claude Bradley, the father of Open Notebook Science, used this photo loop to connect us to some of his activities and give us a glimpse into his personal life.
רפאל נדלן - מה חשוב שתדעו לפני רכישת קרקע?דוד רפאל
דוד רפאל (קבוצת רפאל נדל"ן) הוא יזם נדל"ן .
מעוניינים בקרקע לדירה בתל אביב החל מ-380,000? צרו קשר לגבי פרויקט מגדלי ההשכלה.
כתובת המשרדים: רחוב שהם 5, בנין פז 1 מתחם הבורסה, רמת גן, קומה 19.
פרטי קשר - דוד רפאל - קבוצת רפאל נדל"ן :
טל: 073-7055771 / 8058*
פקס: 073-7055777
International Journal of Engineering Research and Applications (IJERA) is an open access online peer reviewed international journal that publishes research and review articles in the fields of Computer Science, Neural Networks, Electrical Engineering, Software Engineering, Information Technology, Mechanical Engineering, Chemical Engineering, Plastic Engineering, Food Technology, Textile Engineering, Nano Technology & science, Power Electronics, Electronics & Communication Engineering, Computational mathematics, Image processing, Civil Engineering, Structural Engineering, Environmental Engineering, VLSI Testing & Low Power VLSI Design etc.
International Journal of Engineering Research and Applications (IJERA) is an open access online peer reviewed international journal that publishes research and review articles in the fields of Computer Science, Neural Networks, Electrical Engineering, Software Engineering, Information Technology, Mechanical Engineering, Chemical Engineering, Plastic Engineering, Food Technology, Textile Engineering, Nano Technology & science, Power Electronics, Electronics & Communication Engineering, Computational mathematics, Image processing, Civil Engineering, Structural Engineering, Environmental Engineering, VLSI Testing & Low Power VLSI Design etc.
International Journal of Engineering Research and Applications (IJERA) is an open access online peer reviewed international journal that publishes research and review articles in the fields of Computer Science, Neural Networks, Electrical Engineering, Software Engineering, Information Technology, Mechanical Engineering, Chemical Engineering, Plastic Engineering, Food Technology, Textile Engineering, Nano Technology & science, Power Electronics, Electronics & Communication Engineering, Computational mathematics, Image processing, Civil Engineering, Structural Engineering, Environmental Engineering, VLSI Testing & Low Power VLSI Design etc.
International Journal of Engineering Research and Applications (IJERA) is an open access online peer reviewed international journal that publishes research and review articles in the fields of Computer Science, Neural Networks, Electrical Engineering, Software Engineering, Information Technology, Mechanical Engineering, Chemical Engineering, Plastic Engineering, Food Technology, Textile Engineering, Nano Technology & science, Power Electronics, Electronics & Communication Engineering, Computational mathematics, Image processing, Civil Engineering, Structural Engineering, Environmental Engineering, VLSI Testing & Low Power VLSI Design etc.
Single Channel Speech Enhancement using Wiener Filter and Compressive Sensing IJECEIAES
The speech enhancement algorithms are utilized to overcome multiple limitation factors in recent applications such as mobile phone and communication channel. The challenges focus on corrupted speech solution between noise reduction and signal distortion. We used a modified Wiener filter and compressive sensing (CS) to investigate and evaluate the improvement of speech quality. This new method adapted noise estimation and Wiener filter gain function in which to increase weight amplitude spectrum and improve mitigation of interested signals. The CS is then applied using the gradient projection for sparse reconstruction (GPSR) technique as a study system to empirically investigate the interactive effects of the corrupted noise and obtain better perceptual improvement aspects to listener fatigue with noiseless reduction conditions. The proposed algorithm shows an enhancement in testing performance evaluation of objective assessment tests outperform compared to other conventional algorithms at various noise type conditions of 0, 5, 10, 15 dB SNRs. Therefore, the proposed algorithm significantly achieved the speech quality improvement and efficiently obtained higher performance resulting in better noise reduction compare to other conventional algorithms.
Development of Algorithm for Voice Operated Switch for Digital Audio Control ...IJMER
International Journal of Modern Engineering Research (IJMER) is Peer reviewed, online Journal. It serves as an international archival forum of scholarly research related to engineering and science education.
Audio compression has become one of the basic technologies of the multimedia age. The change in the telecommunication infrastructure, in recent years, from circuit switched to packet switched systems has also reflected on the way that speech and audio signals are carried in present systems. In many applications, such as the design of multimedia workstations and high quality audio transmission and storage, the goal is to achieve transparent coding of audio and speech signals at the lowest possible data rates. In other words, bandwidth cost money, therefore, the transmission and storage of information becomes costly. However, if we can use less data, both transmission and storage become cheaper. Further reduction in bit rate is an attractive proposition in applications like remote broadcast lines, studio links, satellite transmission of high quality audio and voice over internet.
METHOD FOR REDUCING OF NOISE BY IMPROVING SIGNAL-TO-NOISE-RATIO IN WIRELESS LANIJNSA Journal
The signal to noise ratio (SNR) is one of the important measures for reducing the noise.A technique that uses a linear prediction error filter (LPEF) and an adaptive digital filter (ADF) to achieve noise reduction in a speech and image degraded by additive background noise is proposed. Since a speech signal can be represented as the stationary signal over a short interval of time, most of speech signal can be predicted by the LPEF. This estimation is performed by the ADF which is used as system identification. Noise reduction is achieved by subtracting the reconstructed noise from the speech degraded by additive background noise. Most of the MR image accelerating methods suffers from degradation of acquired images, which is often correlated with the degree of acceleration. However, Wideband MRI is a novel technique that transcends such flaws.In this paper we proposed LPEF and ADF for reducing the noise in speech and also we demonstrate that Wideband MRI is capable of obtaining images with identical quality as conventional MR images in terms of SNR in wireless LAN.
This paper describe the morphing concept in which we convert the voice of any person into pre -analyzed or pre-recorded voice of any animals.As the user generate a pre-established voice, his pitch, timbre, vibrato and articulation can be modified to resemble those of a pre-recorded and pre-analyzed voice of animal. This technique is based on SMS. Thus using this concept we can develop many funny application and we can used this type of application in mobile device, personal computer etc. for enjoying the sometime of period.
Speech Enhancement Using Spectral Flatness Measure Based Spectral SubtractionIOSRJVSP
This paper is aimed to reduce background noise introduced in speech signal during capture, storage, transmission and processing using Spectral Subtraction algorithm. To consider the fact that colored noise corrupts the speech signal non-uniformly over different frequency bands, Multi-Band Spectral Subtraction (MBSS) approach is exploited wherein amount of noise subtracted from noisy speech signal is decided by a weighting factor. Choice of optimal values of weights decides the performance of the speech enhancement system. In this paper weights are decided based on SFM (Spectral Flatness Measure) than conventional SNR (Signal to Noise Ratio) based rule. Since SFM is able to provide true distinction between speech signal and noise signal. Spectrogram, Mean Opinion Score show that speech enhanced from proposed SFM based MBSS possess better perceptual quality and improved intelligibility than existing SNR based MBSS
Connector Corner: Automate dynamic content and events by pushing a buttonDianaGray10
Here is something new! In our next Connector Corner webinar, we will demonstrate how you can use a single workflow to:
Create a campaign using Mailchimp with merge tags/fields
Send an interactive Slack channel message (using buttons)
Have the message received by managers and peers along with a test email for review
But there’s more:
In a second workflow supporting the same use case, you’ll see:
Your campaign sent to target colleagues for approval
If the “Approve” button is clicked, a Jira/Zendesk ticket is created for the marketing design team
But—if the “Reject” button is pushed, colleagues will be alerted via Slack message
Join us to learn more about this new, human-in-the-loop capability, brought to you by Integration Service connectors.
And...
Speakers:
Akshay Agnihotri, Product Manager
Charlie Greenberg, Host
GDG Cloud Southlake #33: Boule & Rebala: Effective AppSec in SDLC using Deplo...James Anderson
Effective Application Security in Software Delivery lifecycle using Deployment Firewall and DBOM
The modern software delivery process (or the CI/CD process) includes many tools, distributed teams, open-source code, and cloud platforms. Constant focus on speed to release software to market, along with the traditional slow and manual security checks has caused gaps in continuous security as an important piece in the software supply chain. Today organizations feel more susceptible to external and internal cyber threats due to the vast attack surface in their applications supply chain and the lack of end-to-end governance and risk management.
The software team must secure its software delivery process to avoid vulnerability and security breaches. This needs to be achieved with existing tool chains and without extensive rework of the delivery processes. This talk will present strategies and techniques for providing visibility into the true risk of the existing vulnerabilities, preventing the introduction of security issues in the software, resolving vulnerabilities in production environments quickly, and capturing the deployment bill of materials (DBOM).
Speakers:
Bob Boule
Robert Boule is a technology enthusiast with PASSION for technology and making things work along with a knack for helping others understand how things work. He comes with around 20 years of solution engineering experience in application security, software continuous delivery, and SaaS platforms. He is known for his dynamic presentations in CI/CD and application security integrated in software delivery lifecycle.
Gopinath Rebala
Gopinath Rebala is the CTO of OpsMx, where he has overall responsibility for the machine learning and data processing architectures for Secure Software Delivery. Gopi also has a strong connection with our customers, leading design and architecture for strategic implementations. Gopi is a frequent speaker and well-known leader in continuous delivery and integrating security into software delivery.
Builder.ai Founder Sachin Dev Duggal's Strategic Approach to Create an Innova...Ramesh Iyer
In today's fast-changing business world, Companies that adapt and embrace new ideas often need help to keep up with the competition. However, fostering a culture of innovation takes much work. It takes vision, leadership and willingness to take risks in the right proportion. Sachin Dev Duggal, co-founder of Builder.ai, has perfected the art of this balance, creating a company culture where creativity and growth are nurtured at each stage.
Essentials of Automations: Optimizing FME Workflows with ParametersSafe Software
Are you looking to streamline your workflows and boost your projects’ efficiency? Do you find yourself searching for ways to add flexibility and control over your FME workflows? If so, you’re in the right place.
Join us for an insightful dive into the world of FME parameters, a critical element in optimizing workflow efficiency. This webinar marks the beginning of our three-part “Essentials of Automation” series. This first webinar is designed to equip you with the knowledge and skills to utilize parameters effectively: enhancing the flexibility, maintainability, and user control of your FME projects.
Here’s what you’ll gain:
- Essentials of FME Parameters: Understand the pivotal role of parameters, including Reader/Writer, Transformer, User, and FME Flow categories. Discover how they are the key to unlocking automation and optimization within your workflows.
- Practical Applications in FME Form: Delve into key user parameter types including choice, connections, and file URLs. Allow users to control how a workflow runs, making your workflows more reusable. Learn to import values and deliver the best user experience for your workflows while enhancing accuracy.
- Optimization Strategies in FME Flow: Explore the creation and strategic deployment of parameters in FME Flow, including the use of deployment and geometry parameters, to maximize workflow efficiency.
- Pro Tips for Success: Gain insights on parameterizing connections and leveraging new features like Conditional Visibility for clarity and simplicity.
We’ll wrap up with a glimpse into future webinars, followed by a Q&A session to address your specific questions surrounding this topic.
Don’t miss this opportunity to elevate your FME expertise and drive your projects to new heights of efficiency.
Kubernetes & AI - Beauty and the Beast !?! @KCD Istanbul 2024Tobias Schneck
As AI technology is pushing into IT I was wondering myself, as an “infrastructure container kubernetes guy”, how get this fancy AI technology get managed from an infrastructure operational view? Is it possible to apply our lovely cloud native principals as well? What benefit’s both technologies could bring to each other?
Let me take this questions and provide you a short journey through existing deployment models and use cases for AI software. On practical examples, we discuss what cloud/on-premise strategy we may need for applying it to our own infrastructure to get it to work from an enterprise perspective. I want to give an overview about infrastructure requirements and technologies, what could be beneficial or limiting your AI use cases in an enterprise environment. An interactive Demo will give you some insides, what approaches I got already working for real.
Dev Dives: Train smarter, not harder – active learning and UiPath LLMs for do...UiPathCommunity
💥 Speed, accuracy, and scaling – discover the superpowers of GenAI in action with UiPath Document Understanding and Communications Mining™:
See how to accelerate model training and optimize model performance with active learning
Learn about the latest enhancements to out-of-the-box document processing – with little to no training required
Get an exclusive demo of the new family of UiPath LLMs – GenAI models specialized for processing different types of documents and messages
This is a hands-on session specifically designed for automation developers and AI enthusiasts seeking to enhance their knowledge in leveraging the latest intelligent document processing capabilities offered by UiPath.
Speakers:
👨🏫 Andras Palfi, Senior Product Manager, UiPath
👩🏫 Lenka Dulovicova, Product Program Manager, UiPath
Key Trends Shaping the Future of Infrastructure.pdfCheryl Hung
Keynote at DIGIT West Expo, Glasgow on 29 May 2024.
Cheryl Hung, ochery.com
Sr Director, Infrastructure Ecosystem, Arm.
The key trends across hardware, cloud and open-source; exploring how these areas are likely to mature and develop over the short and long-term, and then considering how organisations can position themselves to adapt and thrive.
Accelerate your Kubernetes clusters with Varnish CachingThijs Feryn
A presentation about the usage and availability of Varnish on Kubernetes. This talk explores the capabilities of Varnish caching and shows how to use the Varnish Helm chart to deploy it to Kubernetes.
This presentation was delivered at K8SUG Singapore. See https://feryn.eu/presentations/accelerate-your-kubernetes-clusters-with-varnish-caching-k8sug-singapore-28-2024 for more details.
Generating a custom Ruby SDK for your web service or Rails API using Smithyg2nightmarescribd
Have you ever wanted a Ruby client API to communicate with your web service? Smithy is a protocol-agnostic language for defining services and SDKs. Smithy Ruby is an implementation of Smithy that generates a Ruby SDK using a Smithy model. In this talk, we will explore Smithy and Smithy Ruby to learn how to generate custom feature-rich SDKs that can communicate with any web service, such as a Rails JSON API.
Generating a custom Ruby SDK for your web service or Rails API using Smithy
B45010811
1. Nissim Shewade et al Int. Journal of Engineering Research and Applications www.ijera.com
ISSN : 2248-9622, Vol. 4, Issue 5( Version 1), May 2014, pp.08-11
www.ijera.com 8 | P a g e
A Sound Project Using Python
Dhananjay Sharma, Radheshyam Madge, Rishabh Mishra and Nissim Shewade
Department of Computer Engineering Sinhgad College of Engineering University of Pune, Pune India
Abstract
This paper provides the implementation of various digital audio effects (DAFXs) as a combination of user
defined parameters and input sound signal.An approach to implement various effects like delay based effects,
spatial effects, time varying effects and modulators is provided.A unique listening environment is provided
using 3-D spatialization and localization, simulated surround sound, dialogue normalisation, dynamic range
control and down-mixing.An attempt has also been made for music and voice separation to provide karaoke
effect to the sound.
All the implementations are provided in python which is widely adopted, open source and general purpose
programming language and has a vast array ofcode libraries and development tools, and integrates well with
many other programming languages, frameworks and musical applications.
Keywords- Digital audio effects; Dialog normalization; Dynamic range control; Audio downmixing
I. INTRODUCTION
THIS paper has been written in support of a
sound project which largely encircles the domain of
digital audio effects – DAFX as an acronym. The
Python programming language has been chosen for
development of this project due to its apprehensive
provision of sound API‘s, tools, libraries and support
for audio signal processing and its comprehensive
ability to integrate with other programming
constructs. At an abstract level this project comprises
of two modules - audio effects and dialog
normalization coupled with dynamic range control.
The concept underlying dialog normalization and
dynamic range control goes hand-in-hand. Consider
for example a case of TV channels where each
channel sounds different although all of them have
been set at the same volume. Likewise, a portion of
an audio signal may also sound different from
another one although both portions are played at the
same volume. Hence, the challenge here lies in
controlling and modifying values of chunks (data
packets) in each portion to be not more than, so also
not less than a threshold value, the end result being
both portions sound similar when played at the same
volume. A code for the same has been implemented
in this project.
Digital audio effects (DAFX) are systems
that modify audio signals. These transformations are
made according to some control parameters the
permits and delivers output sounds. The audio effects
module being an extensively vast module in itself can
be further classified into modules characterized by
the inherent property of audio signal that is modified
by each effect as:
Delay based effects – chorus, flanger, vibrato
etc.
Spatial effects – reverberation, panning, 3D
effect for headphones using hrtf
Time varying filters – wah-wah
Modulators – tremolo
Karaoke using audio downmixing
Other exciting effects
In order to reduce the restrictions on the type
of audio file that can be scanned as input, a provision
for audio transcoding (i.e. mp3 to .wav conversion)
and compression of a .wav file has also been
provided for. A dedicated attempt has been made to
create and integrate a comprehensive listening and
sound manipulating environment in a single project
as standalone software which provides for uniqueness
to the project.
II. DIALOG NORMALIZATION AND
DYNAMIC RANGE CONTROL
In accordance with the example of TV
channels that has been mentioned previously in this
paper, in the introduction section we hereby throw
some light on this concept and utilization in sound
based application. The dialog normalization coupled
with dynamic range control algorithm, basically
checks the value of amplitude of each chunk (data
packet) in an audio signal and reduces it to the
threshold value if it is greater than the threshold, and
increases the value to the threshold value if value of
the chunk is less than the threshold. Hence, at the
same volume, all portions of the same audio file
which sounded differently now sound similar. This
RESEARCH ARTICLE OPEN ACCESS
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concept has been implemented using the following
equation:
If x(n)>xh(n),
y(n)=x(n)- δ(n)
And
If x(n)>xl(n),
y(n)=x(n)+ δ(n).
The application of this concept can be found
in the scenario where, a TV channel sound sounds
normal, but advertisement on the same channel sound
louder at the same volume. Using this concept, the
advertisements can be made to sound only as loud as
the TV channel sounds normally.
III. AUDIO EFFECTS
Audio effects is an exciting area in the field
of audio signal processing which throws light on the
technique used to manipulate an audio signal by
modifying one or more of its inherent properties like
time distribution, spatial distribution, frequency etc.
A well-defined comprehensive algorithm has been
used to create each effect only after thorough
reference of many papers published at various
institutions in this regard. Specification of each effect
and the corresponding algorithm used has been
provided in this paper.
Delay-based effects:
These category of audio effects are created
by introducing a time delay in between two samples
of the audio signal. Chorus, flanger and vibrato are
the delay-based effects which have been
implemented in this project.A chorus effect simulates
the effect of multiple instances of the signal which
are close in frequency playing along with the lead
instrument in unison. It is achieved by incorporating
a delay in the given sample using the transition
function δ(n)=x(n-k).The delayed samples of signal
are then transmitted together with the original signal,
i.e. original signal imposed with delayed samples of
original signal.Hence the final equation for a chorus
effect is concluded to be δ(n)=x(n)+x(n-k).
A flanger effect is obtained when two
identical audio signals are mixed with one signal
delayed by a small and gradually changing period.
The time delay here is smaller than 20ms. Peaks and
notches are produced in resultant frequency
spectrum. Feedback is used to enhance the intensity
of peaks and troughs. The algorithm and equation
used to implement a flanger is same as that of chorus
effect. The only difference being a varying delay of
much less magnitude is induced to get a flanger.
Vibrato is a regular fluctuation in pitch,
timbre and/or loudness of the audio signal. An
acoustic signal here, can be visualized as a set of
partials, composing the voiced sound, whose
amplitude and frequency vary along time in a
particular way. In case of vocal vibrato, the position
of these partials, i.e. their frequency variation is
harmonically related to the fundamental frequency
variation and is expected that all of them exhibit the
same regular fluctuation. Vibrato effect is
implemented using the equation δ(n)=x(n)+g.x(n-m).
Here δ(n) represents transition function. Here g is the
gain which retains a constant value between 0.3-0.7.
Also m=t/fs is the sample frequency, y(n) output
signal and x(n) is the input signal.
Spatial Effects:
Reverberation is the persistence of sound in
a particular space after the original sound is
produced. A reverberation, or reverb, is created when
a sound is produced in an enclosed space causing a
large number of echoes to build up and then slowly
decay as the sound is absorbed by the walls and
air. This is most noticeable when the sound source
stops but the reflections continue, decreasing in
amplitude, until they can no longer be heard. In
comparison to a distinct echo that is 50 to 100 ms
after the initial sound, reverberation is many
thousands of echoes that arrive in very quick
succession (.01 – 1 ms between
echoes).Reverberation effect can be implemented
with the help of impulse response.The impulse
response is convulsed with input signal to get the
reverberated output. Discreet convolution formula:
y(n)=∑ x(k)*h(n-k)=x(n)*h(n) (k=-∞ to k=+∞)
Panning is the spread of a sound signal
(either monaural or stereophonic pairs) into a new
stereo or multi-channel sound field.A typical physical
recording console pan control is a knob with a pointer
which can be placed from the 8 o'clock dial position
fully left to the 4 o'clock position fully right. This
software replaces the knob with an on-screen "virtual
knob" or slider for each audio source track which
functions identically to its counterpart on a physical
mix console.The pan control in audio gets its name
from panning action in moving image technology.
The audio pan control can be used in a mix to create
the impression that a source is moving from one side
of the soundstage to the other, although ideally there
would be timing [including phase and Doppler
effects], filtering and reverberation differences
present for a more complete picture of apparent
movement within a defined space. Panning effect has
been implemented using the following equation:
We seek to obtain to signals one for each
Left (L) and Right (R) channel, the gains of which,
gL and gR, are applied to steer the sound across the
stereo audio image. ‘x‘ is mono input. This can be
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achieved by simple 2D rotation, where the angle we
sweep is:
A =[cos(angle), sin(angle); -sin(angle),
cos(angle)]
[gL,gR]‘=A.x
3D audio effect with headphones:
There are two models of implementing this
effect. One is mathematical model and the other is
HRIR convolution method of localizing mono
sources in headphones that could successfully ‗place‘
the source at the desired angle. Each of the methods
had their advantages and disadvantages in
implementation, but the end result seemed to be
reasonably accurate for both. We have implemented
the second one. The HRIR convolution method had
the advantage of not making physical
approximations, but rather measuring the exact
impulse response for the appropriate angle. However,
this method results in discrete locations for all the
angles, and thus a continuous sweep would not be
possible. In its current form, if an angle is not present
in a recorded impulse response, the closest one must
be used.
Time-varying Filters:
Wah-wah (or wa-wa) is an imitative word
(or onomatopoeia) for the sound of altering the
resonance of musical notes to extend expressiveness,
sounding much like a human voice saying the
syllable wah. The wah-wah effect is a spectral glide,
a "modification of the vowel quality of a tone", also
known as a band pass filter. The wah-wah effect is
produced by periodically bringing in and out of play
treble frequencies while a note is sustained. The word
is derived from the sound of the effect itself—in
other words, it is onomatopoeic.This effect has been
implemented using the following equation:
δ(n)=yb(n)+yh(n)
Here, x(n)=input signal, F1 = 2*sin((pi*Fc(1))/Fs),
yb(n)=F1.yh(n)+yb(n-1), yh(n)=x(n)-y1(n-1)-Q1.yb(n-1)
Also, F1 = 2 sin(fc=fs), Q1= 2d and y(n)=output
signal.
Modulators:
Tremolo is basically produces a trembling
effect.Tremolo is a variation in amplitude which
turns the volume of a signal up and down, creating a
"shuddering" effect. This type of effect is often used
by electronic instruments and takes the form of a
multiplication of the sound by a waveform of lower
frequency known as an LFO. The result is similar to
the effect of rapid bowing on a violin or the rapid
keying of a piano. In accordions and related
instruments, tremolo by amplitude modulation is
accomplished through intermodulation betweentwo
or more reeds slightly out of tune with each other.
This effect has been implemented using the following
equation:
trem=(1+ alpha*sin(theta)) ( alpha=constant=0.5)
y = trem*x
Karaoke using Audio Downmixing:
Audio downmixing algorithm essentially
converts a stereo file (having 2 channels) to a mono
file (having 1 channel). The structure and
composition of a wave file attains special
significance in the creation of this effect. All the
vocal content is basically stored in the central part of
a channel in a wave file. Hence, it has been observed
that if two channels of a stereo file are attenuated
separately and then one channel is subtracted from
the other all the vocal content is filtered and all that
remains in the wave file is instrumental music with
only a slight stencil of vocals. Complete elimination
of vocals is an aspect to be considered in the future
scope of this project. Since we directly deal with
channels here, there is no specific equation that
describes implementation of this effect.
Other exciting effects:
In audio engineering, a fade is a gradual
increase or decrease in the level of an audio signal.
The term can also be used for film cinematography.
In fade-out effect the amplitude of a sound decreases
from start to end of sound file. A recorded song may
be gradually reduced to silence at its end. Fading-out
can serve as a recording solution for pieces of music
that contain no obvious ending. In fade-in effect the
amplitude of a sound increases from start to end of
sound file. A recorded song may be gradually
increase from silence at the beginning (fade-in).
The audio denoising effect is used particularly to
remove the white noise present in a signal. During
audio recording there is a possibility that the white
noise present in the background may hamper the
quality of recording. Hence in order to reduce the
white noise denoising is applied. It doesn't reduce the
noise directly to zero, but gradually eliminates it.
The fuzz audio effect comprises of complete non-
linear behaviour, harder/harsher than distortion. It
alters an audio signal until it is nearly a square wave
and adds complex overtones by way of a frequency
multiplier.
In overdrive effect, audio at a low input
level is driven by higher input levels in a non-linear
curve characteristic. Overdrive effects are the mildest
of the three (distortion, fuzz & overdrive), producing
"warm" overtones at quieter volumes and harsher
distortion as gain is increased. A "distortion" effect
produces approximately the same amount of
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distortion at any volume, and its sound alterations are
much more pronounced and intense.
Pitch shifting is a sound recording technique
in which the original pitch of a sound is raised or
lowered. A pitch shifter is a sound effects unit that
raises or lowers the pitch of an audio signal by a
preset interval. For example, a pitch shifter set to
increase the pitch by a fourth will raise each note
three diatonic intervals above the notes actually
played.
Ring modulation is a signal-processing
function in electronics, an implementation of
amplitude modulation or frequency mixing,
performed by multiplying two signals, where one is
typically a sine-wave or another simple waveform. It
is referred to as "ring" modulation because the analog
circuit of diodes originally used to implement this
technique took the shape of a ring. This circuit is
similar to a bridge rectifier, except that instead of the
diodes facing "left" or "right", they go "clockwise" or
"anti-clockwise". A ring modulator is an effects unit
working on this principle.
The robotic effect is widely used in movies
(eg. star wars).The general speech signal is fed with
some carrier wave to add robotic effect. The input is
fed through band-pass filters to separate the tonal
characteristics which then trigger noise generators.
The sounds generated are mixed back with some of
the original sound and this gives the effect.
In rotary speaker effect it appears as if the sound is
rotating. The ―Doppler‖ effect (varying the pitch
based on the speed of moving towards and away from
the sound source) will be familiar to you from
ambulance sirens zooming past in the street! Rotary
effects use the same phenomenon, but on a smaller
scale, and combined with the tremolo (more later)
effect.
The spectral panning effect consists in
converting a mono sound signal into a stereo sound
signal where each frequency component is placed to
its own azimuth position between the loudspeakers,
creating an spectral split effect.
In time stretching you can either stretch or compress
the signal. The time period for which the audio signal
lasts increases or decreases accordingly. It doesn't
change pitch of audio.
IV. CONCLUSION
The audio effects implemented in this
project have been observed to have considerable
accuracy. The karaoke effect has a stencil of vocals
left in the final output which ideally should be
completely eliminated. Thus, complete elimination of
vocals herein should be considered as future scope of
this project. Also optimization in the implementation
of dynamic range control should be considered in the
future scope. All effects promised in this paper have
been implemented and delivered, the accuracy of
each varying to a certain extent.
REFERENCES
[1] UdoZolzer ―DAFX - Digital Audio
Effects‖ Copyright q 2002 John Wiley &
Sons, Ltd ISBNs: 0-471-49078-4
(Hardback); 0-470-84604-6 (Electronic).
[2] IxoneArroabarren, Xavier Rodet, and
Alfonso Carlosena ―On the Measurement of
the Instantaneous Frequency and Amplitude
of Partials in Vocal Vibrato‖IEEE
TRANSACTIONS ON AUDIO, SPEECH,
AND LANGUAGE PROCESSING, VOL.
14, NO. 4, JULY 2006.
[3] UdoZ¨olzer ―PITCH-BASED DIGITAL
AUDIO EFFECTS‖ 5th International
Symposium on Communications, Control
and Signal Processing, ISCCSP 2012,
Rome, Italy, 2-4 May 2012.