The document proposes an algorithm for audio rendering using multiple impulse responses that allows for reproduction of a moving listener position. It analyzes impulse response tails to generate a prototype tail and uses a hybrid reverberation structure including FIR and IIR filters to synthesize the reverberation effect in real-time. Experimental results on a church impulse response database show the approach can accurately reproduce reverberation time and clarity measurements compared to real impulse responses. Informal listening tests found no perceptible differences between the proposed approach and an existing technique.
1) Equalizer matching involves finding the power spectrum of an example audio, then multiplying the input audio's magnitude spectrogram by a filter matching the example's power spectrum.
2) Noise matching involves denoising the input and example separately, then recombining their clean and noise components using the original signal-to-noise ratio.
3) Reverberation matching uses convolutive non-negative matrix factorization to decompose the input into a dry sound and reverb kernel, and convolve the estimated dry input with the example's reverb kernel.
An Advanced Implementation of a Digital Artificial Reverberatora3labdsp
This paper proposes an enhanced hybrid reverberator that uses both measured impulse responses and synthesized impulse responses to model reverberation effects. It develops an automatic procedure to set the parameters of the hybrid reverberator by analyzing the mixing time of measured impulse responses and minimizing a loss function in the cepstral domain. Experimental results show the proposed method produces higher quality reverberation effects than a previous method, with only a small increase in computational cost, as confirmed by listening tests.
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.
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.
This document discusses 3D audio playback through headphones and loudspeakers. It explains that headphones can trick the auditory system but result in sounds inside the head. Loudspeakers externalize sounds but cause crosstalk that must be canceled. Crosstalk cancellation techniques like BACCH aim to reconstruct signals at the ears but can cause spectral coloring. Optimal source distribution improves upon this by varying source positions with frequency. Head tracking further enhances the listening experience for moving listeners.
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.
Ramin Anushiravani presented on example-based audio editing. He discussed techniques for acoustic matching, including manual and automatic equalization, noise reduction, and reverberation matching. For equalization matching, he used a graphic equalizer interface and magnitude spectrogram comparison. Noise matching involved spectral subtraction and post-filtering to reduce musical noise artifacts. Reverberation matching used convolutional NMF to decompose reverberated audio into dry signals and reverb kernels. A user study evaluated participants on matching equalization and reverberation in audio clips.
1) Equalizer matching involves finding the power spectrum of an example audio, then multiplying the input audio's magnitude spectrogram by a filter matching the example's power spectrum.
2) Noise matching involves denoising the input and example separately, then recombining their clean and noise components using the original signal-to-noise ratio.
3) Reverberation matching uses convolutive non-negative matrix factorization to decompose the input into a dry sound and reverb kernel, and convolve the estimated dry input with the example's reverb kernel.
An Advanced Implementation of a Digital Artificial Reverberatora3labdsp
This paper proposes an enhanced hybrid reverberator that uses both measured impulse responses and synthesized impulse responses to model reverberation effects. It develops an automatic procedure to set the parameters of the hybrid reverberator by analyzing the mixing time of measured impulse responses and minimizing a loss function in the cepstral domain. Experimental results show the proposed method produces higher quality reverberation effects than a previous method, with only a small increase in computational cost, as confirmed by listening tests.
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.
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.
This document discusses 3D audio playback through headphones and loudspeakers. It explains that headphones can trick the auditory system but result in sounds inside the head. Loudspeakers externalize sounds but cause crosstalk that must be canceled. Crosstalk cancellation techniques like BACCH aim to reconstruct signals at the ears but can cause spectral coloring. Optimal source distribution improves upon this by varying source positions with frequency. Head tracking further enhances the listening experience for moving listeners.
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.
Ramin Anushiravani presented on example-based audio editing. He discussed techniques for acoustic matching, including manual and automatic equalization, noise reduction, and reverberation matching. For equalization matching, he used a graphic equalizer interface and magnitude spectrogram comparison. Noise matching involved spectral subtraction and post-filtering to reduce musical noise artifacts. Reverberation matching used convolutional NMF to decompose reverberated audio into dry signals and reverb kernels. A user study evaluated participants on matching equalization and reverberation in audio clips.
Ramin Anushiravani's document outlines techniques for sound source localization using microphone arrays. It discusses beamforming methods like delay-and-sum and MVDR beamforming, as well as subspace-based algorithms like MUSIC. It also covers topics like uniform linear arrays, beampatterns, and spatial aliasing. The document presents results from experiments localizing 1-2 sound sources using arrays with 2-4 microphones.
A Hybrid Approach for Real-time Room Acoustic Response Simulationa3labdsp
Reverberation is a well known effect particularly important for music listening especially for recorded and live music. Generally, there are two approaches for artificial reverberation: the desired signal can be obtained by convolving the input signal with a measured impulse response (IR) or a synthetic one. Taking into account the advantages of both approaches, a hybrid artificial reverberation algorithm is presented. The early reflections are derived from a real IR, truncated considering the calculated mixing time, and the reverberation tail is obtained considering the Moorer's structure. The parameters defining this structure are derived from the analyzed IR, using a minimization criteria based on Simultaneous Perturbation Stochastic Approximation (SPSA). The obtained results showed a high-quality reverberator with a low computational load.
This document discusses speech compression using linear predictive coding (LPC). It begins with the objectives of developing low bit-rate speech coders for cellular networks. It then introduces LPC and how it models the human vocal tract. The key aspects of LPC encoding and decoding are described, including analysis, synthesis, and the Levinson-Durbin algorithm. Simulation results on compressing male and female speech are presented, showing compression ratios and signal-to-noise ratios. The document concludes that LPC is well-suited for secure telephone systems by preserving the meaning of speech at low bit rates.
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.
Spatial Fourier transform-based localized sound zone generation with loudspea...Takuma_OKAMOTO
The document describes spatial Fourier transform-based methods for generating localized sound zones with loudspeaker arrays. It discusses previous methods and their problems, and proposes using spatial Fourier transforms to analytically derive spatial filters in the angular spectrum domain. This allows generating bright and dark zones to create localized listening and quiet areas with both linear and circular loudspeaker arrays. Results show the proposed method can generate localized zones more accurately than previous techniques.
DSP_FOEHU - Lec 13 - Digital Signal Processing Applications IAmr E. Mohamed
This document provides an overview of digital signal processing applications including digital spectrum analysis, speech processing, and radar. It discusses different types of digital spectrum analyzers including filter bank, swept, and FFT analyzers. It also covers topics related to speech processing like the anatomy of speech production, speech perception, voiced and unvoiced sounds, and phonemes. Common speech coding techniques are introduced such as vocoding, ADPCM, LPC, and CELP coding. Radar applications of DSP are also briefly mentioned.
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.
Plane wave decomposition and beamforming for directional spatial sound locali...Muhammad Imran
This document describes research on using plane wave decomposition and beamforming with spherical microphone arrays for spatial sound localization. Key points:
- Plane wave decomposition and minimum variance distortionless response beamforming can improve spatial resolution and signal-to-noise ratio for localizing acoustic sources compared to traditional methods.
- A spherical microphone array like the Eigenmike is used to capture sound, which is then decomposed into plane waves using spherical harmonics.
- The methodology is verified in an anechoic chamber with three sound sources localized within 2 degrees of their expected positions.
- Measurements were also conducted in a concert hall to analyze the spatial distribution of the sound field using this method.
Speech is the vocalizer form of human communication,and based upon the syntactic combination of lexical and vocabularies. The aim of speech coding is to compress the speech signal to the highest possible compression ratio bu t maintaining user acceptability.There are many methods for speech compression like Linear Pre dictive coding (LPC),Code Excited Linear Predictive coding (CELP),Sub-band coding,T ransform coding:- Fast Fourier Transform (FFT),Discrete Cosine Transform (DCT),Continuous Wavelet Transform (CWT),Discrete Wavelet Transform (DWT),Variance Fractal Compression (VFC),Discrete Cosine Transform (DCT),Psychoacoustics and etc. Few of them are discus in this paper.
2012 measuring room impulse responses - impact of the decay range on derive...Paulo Abelho
This document investigates the impact of the decay range of room impulse responses on derived room acoustic parameters calculated according to ISO 3382-1. It defines an Impulse Response to Noise Ratio (INR) to estimate the decay range. The study analyzes a large set of practical impulse response measurements to determine the minimum decay range needed to reduce the uncertainty in calculated parameters like reverberation time, clarity, and interaural cross-correlation below the just noticeable difference. The results provide a proposal for specifying the minimum required decay range for each ISO 3382-1 parameter based on the INR and acceptable measurement uncertainty.
This document discusses radio frequency (RF) propagation and link budget analysis. It begins by describing the basic components of a transmission system including the transmitter, propagation path, and receiver. It then covers concepts such as free space path loss, antenna gain, effective isotropic radiated power (EIRP), and the near and far field regions. The document also presents models for calculating path loss in different environments, including the free space and Hata models. It concludes by explaining how link budget analysis can be used to determine the maximum allowable path loss between transmitter and receiver given their power levels, antenna gains, losses, and receiver sensitivity.
Real-time neural text-to-speech with sequence-to-sequence acoustic model and ...Takuma_OKAMOTO
This document proposes a real-time neural text-to-speech system for pitch accent languages using a sequence-to-sequence acoustic model with full-context label input and either a WaveGlow or single Gaussian WaveRNN vocoder. The system realizes high-fidelity synthesis comparable to human speech with a real-time factor of 0.16 using WaveGlow on a GPU. Subjective evaluations show the proposed single Gaussian WaveRNN outperforms other vocoder options. Future work will explore real-time inference on CPUs and compare the sequence-to-sequence acoustic model to conventional pipeline models.
1) The document describes using a Chebyshev filter to remove noise from radar signal data to obtain a clear picture of the radar target track for display. Chebyshev filters have steeper roll-off and more ripple than Butterworth filters but minimize error between the ideal and actual frequency response.
2) The radar signal is passed through a designed 5th order Chebyshev filter with parameters like passband frequency and ripple defined. This significantly increases the signal-to-noise ratio from 10.0085dB to over 10.06dB.
3) The pole-zero plot shows the Chebyshev filter poles lie on an ellipse to minimize frequency response errors over the passband range, with ripp
This document summarizes digital modeling techniques for speech signals. It describes the vocal source and vocal tract that produce speech. It then discusses using sampling and techniques like PCM to digitally represent speech signals. Linear predictive coding is presented as a simple method to analyze speech that approximates samples as combinations of past signals. The summary concludes that linear prediction can be used for spectrum estimation by representing the vocal tract transfer function, pitch detection, and speech synthesis.
Speech Enhancement Based on Spectral Subtraction Involving Magnitude and Phas...IRJET Journal
This document presents a speech enhancement method based on spectral subtraction involving the magnitude and phase components. The proposed method aims to improve noisy speech quality by estimating and subtracting noise from the noisy speech signal in the frequency domain. It involves segmenting the noisy speech into overlapping frames, estimating the noise spectrum during non-speech periods, subtracting the noise magnitude from the noisy speech magnitude, and reconstructing the enhanced speech using the inverse FFT and overlap-add processing. The method was tested on different noise types using MATLAB simulations. Results showed the proposed method achieved better noise reduction compared to conventional spectral subtraction while introducing less speech distortion.
Cancellation of Noise from Speech Signal using Voice Activity Detection Metho...ijsrd.com
Speech Enhancement by suppressing uncorrelated acoustically added noise has been a challenging topic of research for many years. These are the primary choice for real time applications due to the simplicity and comparatively low computational load. This paper shows VAD (Voice activity detection) technique that can detect the non speech segment from the speech signal. It is also shown that it can work powerfully in an unpredictable noise ambience. The technique is mostly done in microprocessors or DSP processors because of their flexibility. But there are several advantages of FPGA over DSP processors like high cost per logic element related to these processors makes them improper for large scale use. From the experimental results, VAD method is implemented on the FPGA chip.
The document discusses color image processing and color models. It describes how color is perceived by the human visual system through rods and cones in the retina. Various color models are examined, including RGB, CMY, HSV, YIQ, and YUV. Color models transform between different representations of color, such as representing a color by its hue, saturation, and intensity rather than red, green, and blue values.
Digital Signal Processing-Digital FiltersNelson Anand
This document discusses digital signal processing using digital filters in MATLAB. It begins by introducing signals and their analog and digital processing. It then covers key digital signal processing tasks like filtering, transforms, and convolution. It describes different filter types including FIR and IIR, and filter design methods. MATLAB sessions are included to demonstrate filtering and filter design. The overall document provides a conceptual overview of digital filters and digital signal processing.
Intelligent Image Enhancement and Restoration - From Prior Driven Model to Ad...Wanjin Yu
ICME2019 Tutorial: Intelligent Image Enhancement and Restoration - From Prior Driven Model to Advanced Deep Learning Part 4: retinex model based low light enhancement
A NOVEL APPROACH TO CHANNEL DECORRELATION FOR STEREO ACOUSTIC ECHO CANCELLATI...a3labdsp
This document proposes a novel approach to decorrelating stereo acoustic signals for acoustic echo cancellation based on the psychoacoustic phenomenon of the "missing fundamental". The approach tracks and removes the pitch from one channel of the stereo signal using an adaptive notch filter, which greatly reduces inter-channel coherence in the lower spectrum without affecting signal quality. Experimental results show the proposed approach provides significant coherence reduction and faster convergence speed of adaptive filters compared to a masked noise injection method, while better preserving the stereo quality.
Low Power High-Performance Computing on the BeagleBoard Platforma3labdsp
The ever increasing energy requirements of supercomputers and server farms is driving the scientific and industrial communities to take in deeper consideration the energy efficiency of computing equipments. This contribution addresses the issue proposing a cluster of ARM processors for high-performance computing. The cluster is composed of five BeagleBoard-xM, with one board managing the cluster, and the other boards executing the actual processing. The software platform is based on the Angstrom GNU/Linux distribution and is equipped with a distributed file system to ease sharing data and code among the nodes of the cluster, and with tools for managing tasks and monitoring the status of each node. The computational capabilities of the cluster have been assessed through High-Performance Linpack and a cluster-wide speaker diarization algorithm, while power consumption has been measured using a clamp meter. Experimental results obtained in the speaker diarization task showed that the energy efficiency of the BeagleBoard-xM cluster is comparable to the one of a laptop computer equipped with a Intel Core2 Duo T8300 running at 2.4 GHz. Furthermore, removing the bottleneck due to the Ethernet interface, the BeagleBoard-xM cluster is able to achieve a superior energy efficiency.
Ramin Anushiravani's document outlines techniques for sound source localization using microphone arrays. It discusses beamforming methods like delay-and-sum and MVDR beamforming, as well as subspace-based algorithms like MUSIC. It also covers topics like uniform linear arrays, beampatterns, and spatial aliasing. The document presents results from experiments localizing 1-2 sound sources using arrays with 2-4 microphones.
A Hybrid Approach for Real-time Room Acoustic Response Simulationa3labdsp
Reverberation is a well known effect particularly important for music listening especially for recorded and live music. Generally, there are two approaches for artificial reverberation: the desired signal can be obtained by convolving the input signal with a measured impulse response (IR) or a synthetic one. Taking into account the advantages of both approaches, a hybrid artificial reverberation algorithm is presented. The early reflections are derived from a real IR, truncated considering the calculated mixing time, and the reverberation tail is obtained considering the Moorer's structure. The parameters defining this structure are derived from the analyzed IR, using a minimization criteria based on Simultaneous Perturbation Stochastic Approximation (SPSA). The obtained results showed a high-quality reverberator with a low computational load.
This document discusses speech compression using linear predictive coding (LPC). It begins with the objectives of developing low bit-rate speech coders for cellular networks. It then introduces LPC and how it models the human vocal tract. The key aspects of LPC encoding and decoding are described, including analysis, synthesis, and the Levinson-Durbin algorithm. Simulation results on compressing male and female speech are presented, showing compression ratios and signal-to-noise ratios. The document concludes that LPC is well-suited for secure telephone systems by preserving the meaning of speech at low bit rates.
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.
Spatial Fourier transform-based localized sound zone generation with loudspea...Takuma_OKAMOTO
The document describes spatial Fourier transform-based methods for generating localized sound zones with loudspeaker arrays. It discusses previous methods and their problems, and proposes using spatial Fourier transforms to analytically derive spatial filters in the angular spectrum domain. This allows generating bright and dark zones to create localized listening and quiet areas with both linear and circular loudspeaker arrays. Results show the proposed method can generate localized zones more accurately than previous techniques.
DSP_FOEHU - Lec 13 - Digital Signal Processing Applications IAmr E. Mohamed
This document provides an overview of digital signal processing applications including digital spectrum analysis, speech processing, and radar. It discusses different types of digital spectrum analyzers including filter bank, swept, and FFT analyzers. It also covers topics related to speech processing like the anatomy of speech production, speech perception, voiced and unvoiced sounds, and phonemes. Common speech coding techniques are introduced such as vocoding, ADPCM, LPC, and CELP coding. Radar applications of DSP are also briefly mentioned.
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.
Plane wave decomposition and beamforming for directional spatial sound locali...Muhammad Imran
This document describes research on using plane wave decomposition and beamforming with spherical microphone arrays for spatial sound localization. Key points:
- Plane wave decomposition and minimum variance distortionless response beamforming can improve spatial resolution and signal-to-noise ratio for localizing acoustic sources compared to traditional methods.
- A spherical microphone array like the Eigenmike is used to capture sound, which is then decomposed into plane waves using spherical harmonics.
- The methodology is verified in an anechoic chamber with three sound sources localized within 2 degrees of their expected positions.
- Measurements were also conducted in a concert hall to analyze the spatial distribution of the sound field using this method.
Speech is the vocalizer form of human communication,and based upon the syntactic combination of lexical and vocabularies. The aim of speech coding is to compress the speech signal to the highest possible compression ratio bu t maintaining user acceptability.There are many methods for speech compression like Linear Pre dictive coding (LPC),Code Excited Linear Predictive coding (CELP),Sub-band coding,T ransform coding:- Fast Fourier Transform (FFT),Discrete Cosine Transform (DCT),Continuous Wavelet Transform (CWT),Discrete Wavelet Transform (DWT),Variance Fractal Compression (VFC),Discrete Cosine Transform (DCT),Psychoacoustics and etc. Few of them are discus in this paper.
2012 measuring room impulse responses - impact of the decay range on derive...Paulo Abelho
This document investigates the impact of the decay range of room impulse responses on derived room acoustic parameters calculated according to ISO 3382-1. It defines an Impulse Response to Noise Ratio (INR) to estimate the decay range. The study analyzes a large set of practical impulse response measurements to determine the minimum decay range needed to reduce the uncertainty in calculated parameters like reverberation time, clarity, and interaural cross-correlation below the just noticeable difference. The results provide a proposal for specifying the minimum required decay range for each ISO 3382-1 parameter based on the INR and acceptable measurement uncertainty.
This document discusses radio frequency (RF) propagation and link budget analysis. It begins by describing the basic components of a transmission system including the transmitter, propagation path, and receiver. It then covers concepts such as free space path loss, antenna gain, effective isotropic radiated power (EIRP), and the near and far field regions. The document also presents models for calculating path loss in different environments, including the free space and Hata models. It concludes by explaining how link budget analysis can be used to determine the maximum allowable path loss between transmitter and receiver given their power levels, antenna gains, losses, and receiver sensitivity.
Real-time neural text-to-speech with sequence-to-sequence acoustic model and ...Takuma_OKAMOTO
This document proposes a real-time neural text-to-speech system for pitch accent languages using a sequence-to-sequence acoustic model with full-context label input and either a WaveGlow or single Gaussian WaveRNN vocoder. The system realizes high-fidelity synthesis comparable to human speech with a real-time factor of 0.16 using WaveGlow on a GPU. Subjective evaluations show the proposed single Gaussian WaveRNN outperforms other vocoder options. Future work will explore real-time inference on CPUs and compare the sequence-to-sequence acoustic model to conventional pipeline models.
1) The document describes using a Chebyshev filter to remove noise from radar signal data to obtain a clear picture of the radar target track for display. Chebyshev filters have steeper roll-off and more ripple than Butterworth filters but minimize error between the ideal and actual frequency response.
2) The radar signal is passed through a designed 5th order Chebyshev filter with parameters like passband frequency and ripple defined. This significantly increases the signal-to-noise ratio from 10.0085dB to over 10.06dB.
3) The pole-zero plot shows the Chebyshev filter poles lie on an ellipse to minimize frequency response errors over the passband range, with ripp
This document summarizes digital modeling techniques for speech signals. It describes the vocal source and vocal tract that produce speech. It then discusses using sampling and techniques like PCM to digitally represent speech signals. Linear predictive coding is presented as a simple method to analyze speech that approximates samples as combinations of past signals. The summary concludes that linear prediction can be used for spectrum estimation by representing the vocal tract transfer function, pitch detection, and speech synthesis.
Speech Enhancement Based on Spectral Subtraction Involving Magnitude and Phas...IRJET Journal
This document presents a speech enhancement method based on spectral subtraction involving the magnitude and phase components. The proposed method aims to improve noisy speech quality by estimating and subtracting noise from the noisy speech signal in the frequency domain. It involves segmenting the noisy speech into overlapping frames, estimating the noise spectrum during non-speech periods, subtracting the noise magnitude from the noisy speech magnitude, and reconstructing the enhanced speech using the inverse FFT and overlap-add processing. The method was tested on different noise types using MATLAB simulations. Results showed the proposed method achieved better noise reduction compared to conventional spectral subtraction while introducing less speech distortion.
Cancellation of Noise from Speech Signal using Voice Activity Detection Metho...ijsrd.com
Speech Enhancement by suppressing uncorrelated acoustically added noise has been a challenging topic of research for many years. These are the primary choice for real time applications due to the simplicity and comparatively low computational load. This paper shows VAD (Voice activity detection) technique that can detect the non speech segment from the speech signal. It is also shown that it can work powerfully in an unpredictable noise ambience. The technique is mostly done in microprocessors or DSP processors because of their flexibility. But there are several advantages of FPGA over DSP processors like high cost per logic element related to these processors makes them improper for large scale use. From the experimental results, VAD method is implemented on the FPGA chip.
The document discusses color image processing and color models. It describes how color is perceived by the human visual system through rods and cones in the retina. Various color models are examined, including RGB, CMY, HSV, YIQ, and YUV. Color models transform between different representations of color, such as representing a color by its hue, saturation, and intensity rather than red, green, and blue values.
Digital Signal Processing-Digital FiltersNelson Anand
This document discusses digital signal processing using digital filters in MATLAB. It begins by introducing signals and their analog and digital processing. It then covers key digital signal processing tasks like filtering, transforms, and convolution. It describes different filter types including FIR and IIR, and filter design methods. MATLAB sessions are included to demonstrate filtering and filter design. The overall document provides a conceptual overview of digital filters and digital signal processing.
Intelligent Image Enhancement and Restoration - From Prior Driven Model to Ad...Wanjin Yu
ICME2019 Tutorial: Intelligent Image Enhancement and Restoration - From Prior Driven Model to Advanced Deep Learning Part 4: retinex model based low light enhancement
A NOVEL APPROACH TO CHANNEL DECORRELATION FOR STEREO ACOUSTIC ECHO CANCELLATI...a3labdsp
This document proposes a novel approach to decorrelating stereo acoustic signals for acoustic echo cancellation based on the psychoacoustic phenomenon of the "missing fundamental". The approach tracks and removes the pitch from one channel of the stereo signal using an adaptive notch filter, which greatly reduces inter-channel coherence in the lower spectrum without affecting signal quality. Experimental results show the proposed approach provides significant coherence reduction and faster convergence speed of adaptive filters compared to a masked noise injection method, while better preserving the stereo quality.
Low Power High-Performance Computing on the BeagleBoard Platforma3labdsp
The ever increasing energy requirements of supercomputers and server farms is driving the scientific and industrial communities to take in deeper consideration the energy efficiency of computing equipments. This contribution addresses the issue proposing a cluster of ARM processors for high-performance computing. The cluster is composed of five BeagleBoard-xM, with one board managing the cluster, and the other boards executing the actual processing. The software platform is based on the Angstrom GNU/Linux distribution and is equipped with a distributed file system to ease sharing data and code among the nodes of the cluster, and with tools for managing tasks and monitoring the status of each node. The computational capabilities of the cluster have been assessed through High-Performance Linpack and a cluster-wide speaker diarization algorithm, while power consumption has been measured using a clamp meter. Experimental results obtained in the speaker diarization task showed that the energy efficiency of the BeagleBoard-xM cluster is comparable to the one of a laptop computer equipped with a Intel Core2 Duo T8300 running at 2.4 GHz. Furthermore, removing the bottleneck due to the Ethernet interface, the BeagleBoard-xM cluster is able to achieve a superior energy efficiency.
Optimized implementation of an innovative digital audio equalizera3labdsp
Digital audio equalization is one of the most common operations in the acoustic field, but its performance
depends on computational complexity and filter design techniques. Starting from a previous FIR
implementation based on multirate systems and filterbanks theory, an optimized digital audio equalizer
is derived. The proposed approach employs IIR filters to improve the filterbanks structure developed to
avoid ripple between adjacent bands. The effectiveness of the optimized implementation is shown comparing
it with the previous approach. The solution presented here has several advantages increasing
the equalization performance in terms of low computational complexity, low delay, and uniform frequency
response.
A Low Latency Implementation of a Non Uniform Partitioned Overlap and Save Al...a3labdsp
FIR convolution is a widely used operation in digital signal processing field, especially for filtering operations in real time scenarios. In this context, low computationally demanding techniques for calculating convolutions with low input/output latency become essential, considering that the real time requirements are strictly related to the impulse response length. In this paper, a multithreading real time implementation of a Non Uniform Partitioned Overlap and Save algorithm is proposed with the aim of lowering the workload required in applications like reverberation, also exploiting the human ear sensitivity. Several results are reported in order to show the effectiveness of the proposed approach in terms of computational cost, taking into consideration different impulse responses and also introducing comparisons with existing techniques of the state of the art.
System Identification Based on Hammerstein Models Using Cubic Splinesa3labdsp
The document presents a new approach for identifying Hammerstein models, which are nonlinear systems composed of a static nonlinearity followed by a linear filter. The approach uses an adaptive Catmull-Rom cubic spline to model the static nonlinearity, instead of high-order polynomials. Experimental results on simulated and real-world systems show the spline-based approach more accurately identifies the nonlinear characteristics and outperforms an existing polynomial-based technique, especially for highly nonlinear systems. The linear filter is modeled using an adaptive IIR filter.
Approximation of Real Impulse Response Using IIR Structures a3labdsp
In this paper, we propose a new approach to the approximation and simulation of a real impulse response. Starting from a preliminary analysis of the mixing time, the impulse response is decomposed in the time domain considering the early and late reflections. Therefore, an IIR structure composed of a cascade of second-order sections and four all-pass filters is employed to synthesize the first part of the impulse response, using a parametric optimization process in the frequency domain. Then, a recursive structure composed of comb and all-pass filters is used to synthesize the late reflections, exploiting a minimization criterion in the cepstral domain. Several results are reported taking into consideration a real impulse response, confirming the validity of the proposed approach.
A Distributed System for Recognizing Home Automation Commands and Distress Ca...a3labdsp
The document describes a distributed system that recognizes home automation commands and distress calls in Italian. It consists of two units: a Local Multimedia Control Unit that recognizes commands/calls and manages communication, and a Central Management Unit that integrates home services and handles emergencies. The system uses acoustic echo cancellation and speech recognition to understand commands even in noisy environments. An evaluation of the system showed it achieved over 90% accuracy on headset microphone data and over 50% on distant microphone data.
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This document summarizes a thesis on developing and implementing an auditory display technique called ADVISE (Auditory Display based on the Virtual Sphere Model).
The document first reviews ADVISE and how it can render spatial audio for both virtual and real scenes using techniques like higher-order ambisonics and binaural rendering. It then reviews the adaptive rectangular decomposition (ARD) method for computational room acoustics simulation.
The document outlines the objective of improving ADVISE by developing a complete room acoustics program using ARD and integrating it into ADVISE. It describes implementing ADVISE to demonstrate spatial audio rendering for a sample music hall scene in Unity. Frequency limitations and computational costs of the
Mixed Time Frequency Approach for Multipoint Room Response Equalizationa3labdsp
A still open problem in the field of room response equalization is the development of perceptually useful mixed-phase equalizers. In a recent paper, a multipoint mixed-phase room response equalization system, integrating a minimum-phase multiple position room magnitude equalizer and a FIR group delay equalizer, was developed in the frequency domain. Starting from this approach, a mixed time-frequency algorithm is here proposed. The minimum-phase multiple position equalizer developed in the frequency domain, is combined with an all-pass FIR phase equalizer, designed in the time domain considering a suitable time-reversed version of a prototype function and taking advantage of the mixing time evaluation. Several tests have been performed considering real environments and comparing the proposed approach with the previous one, based on a group delay compensation. Subjective listening tests have also been done in a real environment, confirming the improvement in the perceived audio quality.
The document discusses spatial hearing and head-related transfer functions (HRTFs) for virtual audio. It covers measuring HRTFs using a KEMAR manikin, constructing filters based on measured HRTFs to localize sound, issues with non-individualized HRTFs, synthetic HRTF approaches, and techniques for externalization like reverberation and decorrelation. Applications mentioned include immersive environments, hearing aids, and representational sounds.
Performance Analysis of Acoustic Echo Cancellation TechniquesIJERA Editor
Mainly, the adaptive filters are implemented in time domain which works efficiently in most of the applications. But in many applications the impulse response becomes too large, which increases the complexity of the adaptive filter beyond a level where it can no longer be implemented efficiently in time domain. An example of where this can happen would be acoustic echo cancellation (AEC) applications. So, there exists an alternative solution i.e. to implement the filters in frequency domain. AEC has so many applications in wide variety of problems in industrial operations, manufacturing and consumer products. Here in this paper, a comparative analysis of different acoustic echo cancellation techniques i.e. Frequency domain adaptive filter (FDAF), Least mean square (LMS), Normalized least mean square (NLMS) &Sign error (SE) is presented. The results are compared with different values of step sizes and the performance of these techniques is measured in terms of Error rate loss enhancement (ERLE), Mean square error (MSE)& Peak signal to noise ratio (PSNR).
Howling occurs when there is an acoustic coupling between a microphone and a speaker, whereby already amplified sound finds its way back into the amplifier through the microphone repeatedly, by making a positive feedback loop that in a way amplifies itself.
This document provides an overview of digital filters and focuses on finite impulse response (FIR) filters. It defines digital filtering and compares it to analog filtering. It describes different types of digital filters including FIR filters and explains how to design, implement and characterize FIR filters. Key aspects of FIR filters are that they have a finite impulse response, linear phase, and are always stable. Design techniques like windowing methods and Parks-McClellan optimization are covered.
A Noise Reduction Method Based on Modified Least Mean Square Algorithm of Rea...IRJET Journal
This document presents a modified least mean square (LMS) algorithm to reduce noise in real-time speech signals. The proposed approach modifies the standard LMS algorithm by incorporating a Wiener filter. Experiments are conducted on speech samples from the NOIZEUS database with various types of noise at different signal-to-noise ratios. Objective measures like segmental SNR, log likelihood ratio, Itakura-Saito spectral distance, and cepstrum are used to evaluate the performance of the proposed algorithm compared to the standard LMS algorithm. The results show that the modified LMS algorithm with Wiener filter outperforms the standard LMS algorithm in enhancing the quality of noisy speech signals based on the objective measure values.
Performance Evaluation of Adaptive Filters Structures for Acoustic Echo Cance...CSCJournals
We have designed and simulated an acoustic echo cancellation system for conferencing. This system is based upon a least-mean-square (LMS) adaptive algorithm and uses multi filter technique. A comparative study of both structure has been carried out and it is found that this new multi-filter converge faster than similar single long adaptive filter. Index Terms: LMS,Multiple sub filter ,Echo cancellation
Room Transfer Function Estimation and Room Equalization in Noise EnvironmentsIJERA Editor
Audio quality in listening situation is degraded by indoor room reverberation. Room equalization can be used to
increase the audio quality by applying the inverse transfer function to the input audio signals. In noise
environments, however, it is hard to exactly measure the room transfer function. In this work, we developed the
techniques to measure the room transfer function in indoor noise environments and to enhance the audio quality
by room equalization. From the experimental results, we showed that the proposed techniques can be
successfully used in indoor noise environments
This document summarizes a research paper on pitch detection of speech synthesis using MATLAB. It discusses using an adaptable filter and peak-valley decision method to determine pitch marks for speech synthesis. Low-pass filtering and autocorrelation are used to detect pitch periods. An adaptive filter is designed to flatten spectral peaks. Peak and valley costs are calculated over each pitch period to determine pitch marks. Dynamic programming is then used to obtain the optimal pitch mark locations for high quality speech synthesis.
This document discusses a study on how the acoustics of a sound control room impact the perceived acoustics of a diffuse field recording played back in that room. The authors used convolution techniques to combine impulse response measurements from concert halls and sound control rooms. They found that for a playback room to accurately reproduce a recording's reverberation time, the playback room should have at least twice the decay rate. For speech intelligibility, the playback room needs a decay rate over four times higher. Initial energy ratios that impact definition and clarity require subjective judgment in the direct sound field. Recommendations for sound control room design by ITU are sufficient for judging reverberation and speech intelligibility of recordings, but clarity needs a much higher
The document discusses different types of digital filters including Infinite Impulse Response (IIR) filters and multirate filters. IIR filters use feedback and have an infinite impulse response. They are potentially unstable but more efficient than FIR filters. IIR filters are usually designed to duplicate analog filter responses and implemented as cascaded second-order sections. Multirate filters involve changing the sample rate, such as decimation which decreases the sample rate, and interpolation which increases the sample rate. Adaptive filters can modify their transfer function based on an optimization algorithm to model non-stationary signals and are used for applications like echo cancellation.
HUFFMAN CODING ALGORITHM BASED ADAPTIVE NOISE CANCELLATIONIRJET Journal
This document presents a paper that proposes using Huffman coding and adaptive noise cancellation algorithms together to reduce noise while transmitting audio and visual signals. It begins with an abstract that discusses using data compression and algorithms to decrease the impact of background noise on recorded signals while maintaining the original undisturbed form of the signals. It then provides background on human hearing capabilities and digital audio signals. The document discusses existing noise cancellation systems and their limitations. It proposes a new framework that uses Huffman coding to create an adaptive code for each unique sound component, and builds a Huffman tree from those codes to map codes to probability. The proposed system is claimed to better remove signal transients and remaining noise artifacts compared to existing short-time Fourier transform approaches.
Depth estimation of sound images using directional clustering and activation-...Daichi Kitamura
Presented at 2014 RISP International Workshop on Nonlinear Circuits, Communications and Signal Processing (NCSP 2014) (international conference)
Tomo Miyauchi, Daichi Kitamura, Hiroshi Saruwatari, Satoshi Nakamura, "Depth estimation of sound images using directional clustering and activation-shared nonnegative matrix factorization," Proceedings of 2014 RISP International Workshop on Nonlinear Circuits, Communications and Signal Processing (NCSP 2014), pp.437-440, Hawaii, USA, March 2014 (Student Paper Award).
Fast Sparse 2-D DFT Computation using Sparse-Graph Alias CodesFrank Ong
This document presents a method called 2D-FFAST (Fast Fourier-Aliasing-based Sparse Transform) that enables fast computation of sparse 2D discrete Fourier transforms (DFTs). It generalizes a previous 1D method to exploit sparsity and allow sub-Nyquist sampling rates. The key ideas are: 1) aliasing patterns from different subsampling reveal sparse entries, 2) choosing co-prime subsampling factors provides diverse patterns, and 3) combining patterns recovers the sparse spectrum. Simulations demonstrate reconstruction of medical images from highly subsampled measurements in both ideal sparse models and more realistic settings.
This document proposes a noise reduction method for audio signals based on an LMS adaptive filter. It segments the noisy audio signal into frames and uses an NLMS adaptive algorithm to estimate the filter coefficients and minimize the mean square error between the clean signal and filter output. Simulation results show the proposed method significantly reduces noise and improves the signal to noise ratio by adaptively filtering the noisy audio signal in the time domain. Analysis of the output signal variance indicates the noise level is substantially decreased compared to the original noisy signal.
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.
Recently, WaveNet, which predicts the probability distribution of speech sample auto-regressively, provides a new paradigm in speech synthesis tasks.
Since the usage of WaveNet for speech synthesis varies by conditional vectors, it is very important to effectively design a baseline system structure.
In this talk, I would like to first introduce various types of WaveNet vocoders such as conventional speech-domain approach and recently proposed source-filter theory-based approach.
Then, I will explain a linear prediction (LP)-based WaveNet speech synthesis, i.e., LP-WaveNet, which overcomes the limitations of source-filter theory-based WaveNet vocoders caused by the mismatch between speech excitation signal and vocal tract filter.
While presenting experimental setups and results, I also would like to share some know-hows to successfully training the network.
Similar to Hybrid Reverberator Using Multiple Impulse Responses for Audio Rendering Improvement (20)
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Nehmen Sie an diesem Webinar teil, bei dem HCL-Ambassador Marc Thomas und Gastredner Franz Walder Ihnen diese neue Welt näherbringen. Es vermittelt Ihnen die Tools und das Know-how, um den Überblick zu bewahren. Sie werden in der Lage sein, Ihre Kosten durch eine optimierte Domino-Konfiguration zu reduzieren und auch in Zukunft gering zu halten.
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Letter and Document Automation for Bonterra Impact Management (fka Social Sol...Jeffrey Haguewood
Sidekick Solutions uses Bonterra Impact Management (fka Social Solutions Apricot) and automation solutions to integrate data for business workflows.
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Interested in deploying letter generation automations for Bonterra Impact Management? Contact us at sales@sidekicksolutionsllc.com to discuss next steps.
Ivanti’s Patch Tuesday breakdown goes beyond patching your applications and brings you the intelligence and guidance needed to prioritize where to focus your attention first. Catch early analysis on our Ivanti blog, then join industry expert Chris Goettl for the Patch Tuesday Webinar Event. There we’ll do a deep dive into each of the bulletins and give guidance on the risks associated with the newly-identified vulnerabilities.
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leewayhertz.com-AI in predictive maintenance Use cases technologies benefits ...alexjohnson7307
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How to Interpret Trends in the Kalyan Rajdhani Mix Chart.pdfChart Kalyan
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Hybrid Reverberator Using Multiple Impulse Responses for Audio Rendering Improvement
1. Audio Rendering Based on Multiple Impulse Responses
Proposed algorithm
Experimental Results
Conclusion
Hybrid Reverberator Using Multiple Impulse
Responses for Audio Rendering Improvement
Andrea Primavera1 , Stefania Cecchi1 , Francesco Piazza1 , Junfeng Li2 , and
Yonghong Yan2
1
A3lab - DII - Universit` Politecnica delle Marche a
Ancona - ITALY
2
Institute of Acoustics, Chinese Academy of
Sciences - Beijing - CHINA
IIHMSP, October 2013, Beijing, China.
Andrea Primavera
Hybrid Reverberator Using Multiple Impulse Responses for Audio Rendering Improvement
1/22
2. Audio Rendering Based on Multiple Impulse Responses
Proposed algorithm
Experimental Results
Conclusion
1
Audio Rendering Based on Multiple Impulse Responses
Introduction
State of the art
Proposed algorithm
2
Proposed algorithm
Analysis of reverberation tail
Synthesis of the reverberation effect
Real-time Reproduction of Moving Listener Position
3
Experimental Results
Experimental Setup
Reverberation Time
Clarity Index
Subjective Analysis
4
Conclusion
Conclusion
Questions
Bibliography
Andrea Primavera
Hybrid Reverberator Using Multiple Impulse Responses for Audio Rendering Improvement
2/22
3. Audio Rendering Based on Multiple Impulse Responses
Proposed algorithm
Experimental Results
Conclusion
Introduction
State of the art
Proposed algorithm
Linear convolution is a widely used operation typically employed for audio
rendering purpose aiming to reproduce the reverberation effect generated
when a sound is produced within an enclosed space.
One of the main problems
LINEAR
CONVOLUTION
STATIC
PROCEDURE
It allows to reproduce only the acoustic effect produced taking into
account a specific sound source with the relative receiver position.
Solution
• Time varying convolution to simulate the moving receiver
positions performing IRs interpolation [1].
• A large impulse response (IR) database is needed.
Andrea Primavera
Hybrid Reverberator Using Multiple Impulse Responses for Audio Rendering Improvement
3/22
4. Audio Rendering Based on Multiple Impulse Responses
Proposed algorithm
Experimental Results
Conclusion
Introduction
State of the art
Proposed algorithm
Linear convolution is a widely used operation typically employed for audio
rendering purpose aiming to reproduce the reverberation effect generated
when a sound is produced within an enclosed space.
One of the main problems
LINEAR
CONVOLUTION
STATIC
PROCEDURE
It allows to reproduce only the acoustic effect produced taking into
account a specific sound source with the relative receiver position.
Solution
• Time varying convolution to simulate the moving receiver
positions performing IRs interpolation [1].
• A large impulse response (IR) database is needed.
Andrea Primavera
Hybrid Reverberator Using Multiple Impulse Responses for Audio Rendering Improvement
3/22
5. Audio Rendering Based on Multiple Impulse Responses
Proposed algorithm
Experimental Results
Conclusion
Introduction
State of the art
Proposed algorithm
Linear convolution is a widely used operation typically employed for audio
rendering purpose aiming to reproduce the reverberation effect generated
when a sound is produced within an enclosed space.
One of the main problems
LINEAR
CONVOLUTION
STATIC
PROCEDURE
It allows to reproduce only the acoustic effect produced taking into
account a specific sound source with the relative receiver position.
Solution
• Time varying convolution to simulate the moving receiver
positions performing IRs interpolation [1].
• A large impulse response (IR) database is needed.
Andrea Primavera
Hybrid Reverberator Using Multiple Impulse Responses for Audio Rendering Improvement
3/22
6. Audio Rendering Based on Multiple Impulse Responses
Proposed algorithm
Experimental Results
Conclusion
Introduction
State of the art
Proposed algorithm
PROBLEM: Large impulse responses database required high memory
usage.
In [2] a database reduction procedure for auralization purpose with moving
listener position has been proposed:
Consideration
• Early reflections contain most of the information regarding the
location of the sound source and receiver.
• Late reflections gives more information about room properties
(e.g., size, geometry, materials) [3].
• The information in the late reverberation tail is largely
redundant across multiple impulse responses recorded in the
same space.
Andrea Primavera
Hybrid Reverberator Using Multiple Impulse Responses for Audio Rendering Improvement
4/22
7. Audio Rendering Based on Multiple Impulse Responses
Proposed algorithm
Experimental Results
Conclusion
Introduction
State of the art
Proposed algorithm
PROBLEM: Large impulse responses database required high memory
usage.
In [2] a database reduction procedure for auralization purpose with moving
listener position has been proposed:
Metodology
• Mixing time evaluation to discriminate late from early
reflections.
• Approximation of the reverberation tail of the whole IR
database as stochastic process (i.e., white noise).
Andrea Primavera
Hybrid Reverberator Using Multiple Impulse Responses for Audio Rendering Improvement
5/22
8. Audio Rendering Based on Multiple Impulse Responses
Proposed algorithm
Experimental Results
Conclusion
Introduction
State of the art
Proposed algorithm
PROBLEM: Large impulse responses database required high memory
usage.
Proposed Solution
Taking into account the procedure described in [2] a novel methodology has been proposed considering the advantages introduced by
hybrid reveberation structure [4] [5].
• It is possible to approximate the convolution operation using
recursive structure (i.e., IIR filters).
• This procedure allows to further reduce the database dimension
with respect to [2].
• The employed structures permits to decrease the computational
load required to perform the real-time auralization.
Andrea Primavera
Hybrid Reverberator Using Multiple Impulse Responses for Audio Rendering Improvement
6/22
9. Audio Rendering Based on Multiple Impulse Responses
Proposed algorithm
Experimental Results
Conclusion
Analysis of reverberation tail
Synthesis of the reverberation effect
Real-time Reproduction of Moving Listener Position
Three are the main phases of the approach presented for the reproduction
of moving listener position exploiting a hybrid reverberator structure:
1
2
3
Andrea Primavera
Analysis of reverberation tail:
Generate a prototype representing the database average reveberation
tail.
Synthesis of the reverberation effect:
Approximate the reverberation tail prototype exploiting an hybrid
reverberation algorithm [4] [5].
Real-time reproduction of moving listener position:
Reverberation effect reproduction using the hybrid reverberation
structure (mixed FIR/IIR filter network).
Hybrid Reverberator Using Multiple Impulse Responses for Audio Rendering Improvement
7/22
10. Audio Rendering Based on Multiple Impulse Responses
Proposed algorithm
Experimental Results
Conclusion
1
Analysis of reverberation tail
Synthesis of the reverberation effect
Real-time Reproduction of Moving Listener Position
Analysis of reverberation tail:
• Mixing time analysis:
The partitioning of early from late reflections has been performed
exploiting gaussianity [4] [6] and phase evolution estimators [7].
• Prototype evaluation:
The reverberation tail prototype is computed as a mean of the IRs
database after the maximum mixing time tm .
htail =
1
N
Lm
hn (t)
(1)
hn : database IRs
t=tm
• Scaling operation:
In order to simulate the distance among the source and the different
listereners position a scaling factor is evaluated.
Sm
Andrea Primavera
RMS(hs )
=
=
RMS(hm )
1
Lm
1
Lm
Lm
t=tm
Lm
t=tm
2
hs (t)
(2)
hs : synthesized IR
hm : original IR
2
hm (t)
Hybrid Reverberator Using Multiple Impulse Responses for Audio Rendering Improvement
8/22
11. Audio Rendering Based on Multiple Impulse Responses
Proposed algorithm
Experimental Results
Conclusion
2
Analysis of reverberation tail
Synthesis of the reverberation effect
Real-time Reproduction of Moving Listener Position
Synthesis of the reverberation effect:
gain
x[n]
EARLY
REFLECTIONS
DEVICE
LATE
REFLECTIONS
DEVICE
+
×
DELAY
y[n]
Hybrid reverberator block diagram for the single channel case.
Early reflections device
Based on the convolution with a
real IR for the reproduction of the
early echoes.
A
+
LBCF
+
LBCF
y[n]
x[n]
Late reflections device
+
+
LBCF
+
Based on IIR filters network (e.g.,
comb and/or all-pass) and a FDN
matrix [8] for the simulation of the
reverberation tail.
Andrea Primavera
AP
AP
+
NAP filters
LBCF
NLBCF filters
Late reflections device block diagram
for the single channel case.
Hybrid Reverberator Using Multiple Impulse Responses for Audio Rendering Improvement
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12. Audio Rendering Based on Multiple Impulse Responses
Proposed algorithm
Experimental Results
Conclusion
2
Analysis of reverberation tail
Synthesis of the reverberation effect
Real-time Reproduction of Moving Listener Position
Synthesis of the reverberation effect:
gain
x[n]
EARLY
REFLECTIONS
DEVICE
LATE
REFLECTIONS
DEVICE
+
×
DELAY
y[n]
Hybrid reverberator block diagram for the single channel case.
Autotuning procedure
An automatic procedure allows to
set the parameters of hybrid reverberator in order to emulate a
real environment starting from its
impulse.
A
+
LBCF
+
LBCF
y[n]
x[n]
+
+
LBCF
+
AP
AP
+
NAP filters
LBCF
NLBCF filters
FIR TO IIR APPROXIMATION
Andrea Primavera
Late reflections device block diagram
for the single channel case.
Hybrid Reverberator Using Multiple Impulse Responses for Audio Rendering Improvement
10/22
13. Audio Rendering Based on Multiple Impulse Responses
Proposed algorithm
Experimental Results
Conclusion
2
Analysis of reverberation tail
Synthesis of the reverberation effect
Real-time Reproduction of Moving Listener Position
Synthesis of the reverberation effect:
Two are the main phases of the autotuning procedure:
Late Reflections Analysis
Early Reflections Partitioning
Evaluation of the mixing time to set the
early reflection device:
•
•
Andrea Primavera
Gaussianity estimators:
Similarities between IR behavior
and gaussian noise can be found
in late reflections.
Kurtosis and MAD/SD ratio
have been used.
Phase distortion evaluation:
The unwrapped phase of the IR
tends to become not linear with
late reflections evolution.
An offline adaptation procedure, based on SPSA [9], has been used to
iteratively find the IIR parameters.
A single loss function computed in cepstral domain [10] has been adopted in
the minimization procedure.
L = max
max
[Tr (i, j) − Ta (i, j)]2
i=1 j=1
K
M
where:
• Tr is a matrix representing the
MFCC derived from the real IR.
•
Ta is the MFCC obtained by the
artificial IR.
Hybrid Reverberator Using Multiple Impulse Responses for Audio Rendering Improvement
11/22
14. Audio Rendering Based on Multiple Impulse Responses
Proposed algorithm
Experimental Results
Conclusion
3
Analysis of reverberation tail
Synthesis of the reverberation effect
Real-time Reproduction of Moving Listener Position
Real-time reproduction of moving listener position:
The movement reproduction is generate changing the reverberator parameters as a function of the listener position:
• Early reflection device: The filter coefficientsare obtained as a
linear or bilinear interpolation of the impulse response, with relation
to the number of the involved IR.
• Late reflection device: The coefficients are fixed reproducing the
same reverberation tail for all the different positions.
Andrea Primavera
Hybrid Reverberator Using Multiple Impulse Responses for Audio Rendering Improvement
12/22
15. Audio Rendering Based on Multiple Impulse Responses
Proposed algorithm
Experimental Results
Conclusion
Experimental Setup
Reverberation Time
Clarity Index
Subjective Analysis
The effectiveness of the presented technique has been proved taking into
account account the IR database of a real environment (i.e., St. Margarets
Church in York [11]).
As reported in [11], a total of 18
IRs has been derived using:
• A logarithmic sweep signal
excitation (20 Hz - 22 kHz).
• Sample rate of 96kHz.
• A Soundfield SPS422B
microphone.
• A Genelec S30D loudspeaker.
Andrea Primavera
Hybrid Reverberator Using Multiple Impulse Responses for Audio Rendering Improvement
13/22
16. Audio Rendering Based on Multiple Impulse Responses
Proposed algorithm
Experimental Results
Conclusion
Experimental Setup
Reverberation Time
Clarity Index
Subjective Analysis
The reverberation time as a function of frequency has been analyzed in
order to provide an objective evaluation between real and synthesize IRs.
Energy Decay Relief (EDR) of one of
the eighteen real IR.
Andrea Primavera
Energy Decay Relief (EDR) of
artificial IR.
Hybrid Reverberator Using Multiple Impulse Responses for Audio Rendering Improvement
14/22
17. Audio Rendering Based on Multiple Impulse Responses
Proposed algorithm
Experimental Results
Conclusion
Experimental Setup
Reverberation Time
Clarity Index
Subjective Analysis
The reverberation time as a function of frequency has been analyzed in
order to provide an objective evaluation between real and synthesize IRs.
Mean difference in reverberation time between the measured and synthesized IRs
exploiting (a) the proposed approach and (b) the technique described in [2].
Since the obtained errors are comparable, the effectiveness of the proposed
technique in time frequency behaviors reproduction is confirmed.
Andrea Primavera
Hybrid Reverberator Using Multiple Impulse Responses for Audio Rendering Improvement
15/22
18. Audio Rendering Based on Multiple Impulse Responses
Proposed algorithm
Experimental Results
Conclusion
Experimental Setup
Reverberation Time
Clarity Index
Subjective Analysis
Another parameter employed in objective analysis is clarity (C50 and C80):
Proposed approach
Approach presented in [2]
Mean real
1.75
1.75
Proposed approach
Approach presented in [2]
4.23
4.23
C50
Mean synth
2.28
0.10
C80
5.49
3.41
Mean err
0.52
1.65
Std err
1.09
1.97
1.26
0.81
1.06
1.29
Clarity measures: mean and the standard deviation (STD) error computed as
difference in corresponding receiver positions of the synthesized and measured IRs
exploiting the proposed approach and the one described in [2].
The similar values obtained as mean and standard deviation evaluation
confirms the validity of the approach.
Andrea Primavera
Hybrid Reverberator Using Multiple Impulse Responses for Audio Rendering Improvement
16/22
19. Audio Rendering Based on Multiple Impulse Responses
Proposed algorithm
Experimental Results
Conclusion
Experimental Setup
Reverberation Time
Clarity Index
Subjective Analysis
Informal listening tests have been performed:
• The movement of the listener position along one dimension has been
simulated.
• The effectiveness of the approach has been confirmed since listeners
are not able to hear any difference between the presented approach
and the one described in [2].
Andrea Primavera
Hybrid Reverberator Using Multiple Impulse Responses for Audio Rendering Improvement
17/22
20. Audio Rendering Based on Multiple Impulse Responses
Proposed algorithm
Experimental Results
Conclusion
Conclusion
Questions
Bibliography
In conclusion:
• A novel approach for the reproduction of moving listener position
exploiting time variant hybrid reverberation algorithm has been
presented.
• As confirmed in several papers the employment of IIR filter network
for the approximation of convolution operation allows to reduce the
computational cost required in the auralization operation, moreover
the approach also allow to decrease the IR database size reducing
the information required for the late reflection reprodution.
• The effectiveness of the approach has been proved taking into
account a real IR database providing comparison with the existing
state-of-art techniques in terms of objective and subjective measures.
Andrea Primavera
Hybrid Reverberator Using Multiple Impulse Responses for Audio Rendering Improvement
18/22
21. Audio Rendering Based on Multiple Impulse Responses
Proposed algorithm
Experimental Results
Conclusion
Conclusion
Questions
Bibliography
QUESTIONS?
Andrea Primavera
Hybrid Reverberator Using Multiple Impulse Responses for Audio Rendering Improvement
19/22
22. Audio Rendering Based on Multiple Impulse Responses
Proposed algorithm
Experimental Results
Conclusion
Conclusion
Questions
Bibliography
B. Dalenback and M. Stromberg, “Real time walk-through
auralization - the first year,” in Proc. Institute of Acoustics,
Amsterdam, NL, Mar. 2006.
R. Stewart and M. Sandler, “Generating a spatial average
reverberation tail across multiple impulse responses,” in Proc. 35th
Audio Engineering Society Conference, London, UK, Dec. 2009.
B. Blesser, “An interdisciplinary synthesis of reverberation
viewpoints,” J. Audio Eng. Soc., vol. 49, no. 10, pp. 867–903, Oct.
2001.
A. Primavera, S. Cecchi, P. Peretti, L. Romoli, and F. Piazza, “An
Advanced Implementation of a Digital Artificial Reverberator,” in
Proc. 130th Audio Engineering Society Convention, London,UK,
May 2011.
A. Primavera, M. Gasparini, S. Cecchi, L. Romoli, and F. Piazza,
“Hybrid Reverberation Algorithm: a Practical Approach,” in
AIA-DAGA Conference, Merano, Italy, Mar. 2013.
Andrea Primavera
Hybrid Reverberator Using Multiple Impulse Responses for Audio Rendering Improvement
20/22
23. Audio Rendering Based on Multiple Impulse Responses
Proposed algorithm
Experimental Results
Conclusion
Conclusion
Questions
Bibliography
R. Stewart and M. Sandler, “Statisical measures of early reections of
room impulse responses,” in in DAFX 07), Bordeaux, France, Sep.
2007.
G. Defrance and J. Polack, “Measuring the mixing time in
auditoria,” in Proc. 155th Meeting of the Acoustical Society of
America), vol. 49, Jun 2001, pp. 867–903.
J. Jot, “Digital Delay Networks for designing artificial reverberators,”
in Proc. 90th Audio Engineering Society Convention, Paris, Feb
1991.
J. Spall, “Implementation of the Simultaneous Perturbation
Algorithm for Stochastic Optimization,” in IEEE Transactions on
Aerospace and Electronic Systems, vol. 34, 1998, pp. 817–823.
S. Heise, M. Hlatky, and J. Loviscach, “Automatic Adjustment of
Off-the-Shelf Reverberation Effects,” in Proc. 126th Audio
Engineering Society Convention, Munich, Germany, May 2009.
Andrea Primavera
Hybrid Reverberator Using Multiple Impulse Responses for Audio Rendering Improvement
21/22
24. Audio Rendering Based on Multiple Impulse Responses
Proposed algorithm
Experimental Results
Conclusion
Conclusion
Questions
Bibliography
“OpenAIR, Audiolab, University of York.” [Online]. Available:
http://www.openairlib.net/
Andrea Primavera
Hybrid Reverberator Using Multiple Impulse Responses for Audio Rendering Improvement
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