Experiments on DCASE 2016: Acoustic Scene Classification and Sound Events Detection in Real life recordings.
Presented at DCASE 2016 Workshop. Among one of the top performing teams in two tasks - task 1 and task 4 for DCASE
Empirical Evaluation of Decomposition Strategy for Wavelet Video CompressionCSCJournals
Abstract The wavelet transform has become the most interesting new algorithm for video compression. Yet there are many parameters within a wavelet analysis and synthesis which govern the quality of a decoded video. In this paper different wavelet decomposition strategies and their implications for the decoded video are discussed. A pool of color video sequences has been wavelet-transformed at different settings of the wavelet filter bank and quantization threshold and with decomposition of dyadic and packet wavelet transformation strategies. The empirical evaluation of the decomposition strategy is based on three benchmarks: a first judgment regards the perceived quality of the decoded video. The compression rate is a second crucial factor, and finally the best parameter setting with regards to the Peak Signal to Noise Ratio (PSNR). The investigation proposes dyadic decomposition as the chosen decomposition strategy.
Video Denoising using Transform Domain MethodIRJET Journal
This document presents a proposed method for video denoising using dictionary learning and transform domain techniques. It begins with an abstract describing how traditional video denoising models based on Gaussian noise do not account for real-world noise sources. The proposed method then learns basis functions adaptively from input video frames using dictionary learning, providing a sparse representation. Hard thresholding is applied in the transform domain to compute denoised frames. Experimental results on standard test videos show the method achieves competitive performance compared to other approaches in terms of peak signal-to-noise ratio.
comparision of lossy and lossless image compression using various algorithmchezhiyan chezhiyan
This document compares lossy and lossless image compression using various algorithms. It discusses the need for image compression to reduce file sizes for storage and transmission. Lossy compression provides higher compression ratios but some loss of information, while lossless compression retains all information without loss. The document proposes comparing algorithms like Fractal image compression and LZW, analyzing parameters like SNR, PSNR, and MSE for formats like BMP, TIFF, PNG and JPEG. It provides details on how the LZW and Fractal compression algorithms work.
The document discusses multimedia and multimedia applications. It covers:
1. The differences between multimedia applications and classic applications in terms of delay sensitivity and packet loss tolerance.
2. The main classes of multimedia applications including streaming stored audio/video, streaming live audio/video, and real-time interactive audio/video. It outlines the requirements and constraints of each class.
3. Problems with delivering multimedia over the Internet like limited bandwidth, packet jitter, and packet loss. It describes solutions to these problems including compression techniques, playout buffering, and forward error correction.
4. Common multimedia protocols including RTP for framing/multiplexing/synchronization and RTCP for feedback. It explains how these
Compression: Video Compression (MPEG and others)danishrafiq
This document provides an overview of video compression techniques used in standards like MPEG and H.261. It discusses how uncompressed video data requires huge storage and bandwidth that compression aims to address. It explains that lossy compression methods are needed to achieve sufficient compression ratios. The key techniques discussed are intra-frame coding using DCT and quantization similar to JPEG, and inter-frame coding using motion estimation and compensation to remove temporal redundancy between frames. Motion vectors are found using techniques like block matching and sum of absolute differences. MPEG and other standards use a combination of these intra and inter-frame coding techniques to efficiently compress video for storage and transmission.
The document discusses multimedia compression techniques. It notes that audio, image, and video files require large amounts of data, which bandwidth and storage limitations cannot accommodate for real-time transmission and playback. Compression reduces these file sizes through lossless and lossy techniques. Popular standards like JPEG and MPEG use combinations of techniques like the discrete cosine transform, quantization, predictive coding, and entropy encoding to achieve compression while maintaining quality.
Introduction to Video Compression Techniques - Anurag JainVideoguy
The document provides an overview of video compression techniques and standards. It discusses the motivation for video compression to reduce data sizes for storage and transmission. It then reviews several key video compression standards including H.261, H.263, MPEG-1, MPEG-2, MPEG-4, H.264 and others. For each standard, it summarizes the goals, features, applications and technical details like motion compensation methods, block sizes, and bitrate ranges.
Deep Learning Based Voice Activity Detection and Speech EnhancementNAVER Engineering
The document summarizes speech recognition front-end technologies including voice activity detection (VAD) and speech enhancement. It discusses conventional signal processing based approaches and more recent deep learning based methods. For VAD, it describes adaptive context attention models that can dynamically adjust the context used based on noise type and SNR. For speech enhancement, it proposes a two-step neural network approach consisting of a prior network that makes multiple predictions from noisy features and a post network that combines these using a boosting method to produce enhanced features, allowing end-to-end training without an explicit masking step. The approach aims to better exploit neural network modeling power while reducing computation cost compared to conventional methods or single-step deep learning frameworks.
Empirical Evaluation of Decomposition Strategy for Wavelet Video CompressionCSCJournals
Abstract The wavelet transform has become the most interesting new algorithm for video compression. Yet there are many parameters within a wavelet analysis and synthesis which govern the quality of a decoded video. In this paper different wavelet decomposition strategies and their implications for the decoded video are discussed. A pool of color video sequences has been wavelet-transformed at different settings of the wavelet filter bank and quantization threshold and with decomposition of dyadic and packet wavelet transformation strategies. The empirical evaluation of the decomposition strategy is based on three benchmarks: a first judgment regards the perceived quality of the decoded video. The compression rate is a second crucial factor, and finally the best parameter setting with regards to the Peak Signal to Noise Ratio (PSNR). The investigation proposes dyadic decomposition as the chosen decomposition strategy.
Video Denoising using Transform Domain MethodIRJET Journal
This document presents a proposed method for video denoising using dictionary learning and transform domain techniques. It begins with an abstract describing how traditional video denoising models based on Gaussian noise do not account for real-world noise sources. The proposed method then learns basis functions adaptively from input video frames using dictionary learning, providing a sparse representation. Hard thresholding is applied in the transform domain to compute denoised frames. Experimental results on standard test videos show the method achieves competitive performance compared to other approaches in terms of peak signal-to-noise ratio.
comparision of lossy and lossless image compression using various algorithmchezhiyan chezhiyan
This document compares lossy and lossless image compression using various algorithms. It discusses the need for image compression to reduce file sizes for storage and transmission. Lossy compression provides higher compression ratios but some loss of information, while lossless compression retains all information without loss. The document proposes comparing algorithms like Fractal image compression and LZW, analyzing parameters like SNR, PSNR, and MSE for formats like BMP, TIFF, PNG and JPEG. It provides details on how the LZW and Fractal compression algorithms work.
The document discusses multimedia and multimedia applications. It covers:
1. The differences between multimedia applications and classic applications in terms of delay sensitivity and packet loss tolerance.
2. The main classes of multimedia applications including streaming stored audio/video, streaming live audio/video, and real-time interactive audio/video. It outlines the requirements and constraints of each class.
3. Problems with delivering multimedia over the Internet like limited bandwidth, packet jitter, and packet loss. It describes solutions to these problems including compression techniques, playout buffering, and forward error correction.
4. Common multimedia protocols including RTP for framing/multiplexing/synchronization and RTCP for feedback. It explains how these
Compression: Video Compression (MPEG and others)danishrafiq
This document provides an overview of video compression techniques used in standards like MPEG and H.261. It discusses how uncompressed video data requires huge storage and bandwidth that compression aims to address. It explains that lossy compression methods are needed to achieve sufficient compression ratios. The key techniques discussed are intra-frame coding using DCT and quantization similar to JPEG, and inter-frame coding using motion estimation and compensation to remove temporal redundancy between frames. Motion vectors are found using techniques like block matching and sum of absolute differences. MPEG and other standards use a combination of these intra and inter-frame coding techniques to efficiently compress video for storage and transmission.
The document discusses multimedia compression techniques. It notes that audio, image, and video files require large amounts of data, which bandwidth and storage limitations cannot accommodate for real-time transmission and playback. Compression reduces these file sizes through lossless and lossy techniques. Popular standards like JPEG and MPEG use combinations of techniques like the discrete cosine transform, quantization, predictive coding, and entropy encoding to achieve compression while maintaining quality.
Introduction to Video Compression Techniques - Anurag JainVideoguy
The document provides an overview of video compression techniques and standards. It discusses the motivation for video compression to reduce data sizes for storage and transmission. It then reviews several key video compression standards including H.261, H.263, MPEG-1, MPEG-2, MPEG-4, H.264 and others. For each standard, it summarizes the goals, features, applications and technical details like motion compensation methods, block sizes, and bitrate ranges.
Deep Learning Based Voice Activity Detection and Speech EnhancementNAVER Engineering
The document summarizes speech recognition front-end technologies including voice activity detection (VAD) and speech enhancement. It discusses conventional signal processing based approaches and more recent deep learning based methods. For VAD, it describes adaptive context attention models that can dynamically adjust the context used based on noise type and SNR. For speech enhancement, it proposes a two-step neural network approach consisting of a prior network that makes multiple predictions from noisy features and a post network that combines these using a boosting method to produce enhanced features, allowing end-to-end training without an explicit masking step. The approach aims to better exploit neural network modeling power while reducing computation cost compared to conventional methods or single-step deep learning frameworks.
Sparse Approximation of Gram Matrices for GMMN-based Speech SynthesisTomoki Koriyama
The document examines approximation methods for reducing the computational cost of training generative moment matching network (GMMN)-based speech synthesis. It investigates approximating the Gram matrices using block diagonal and random Fourier feature approaches, as well as using random and clustering-based methods for minibatch selection. Subjective tests found that the random Fourier feature and clustering-based minibatch selection approaches achieved higher subjective scores for inter-utterance variation while maintaining naturalness.
This document summarizes video compression techniques used in standards like MPEG. It discusses how video contains redundancies that compression reduces by encoding differences between frames and subsampling color. The MPEG algorithm compresses video in 5 steps: resolution reduction, motion estimation between frame types (I, P, B frames), discrete cosine transform, quantization, and entropy encoding. Standards like MPEG-1, 2, and 4 provide different compression ratios and capabilities for various applications like streaming video.
The document discusses video compression basics and MPEG-2 video compression. It explains that video frames contain redundant spatial and temporal data that can be compressed. MPEG-2 uses three frame types (I, P, B frames) and compresses frames using intra-frame and inter-frame encoding techniques like DCT, quantization, and entropy encoding to remove redundancy. The encoding process transforms raw video frames to compressed bitstreams for efficient storage and transmission.
Signal and image processing on satellite communication using MATLABEmbedded Plus Trichy
Basic Explanations about satellite imaging and signal processing with the help of MATLAB.
Contact us: 23,Nandhi koil Street, Near Nakoda Showroom,Theppakulam,Trichy
Mb.No:9360212155.
Mail:embeddedplusproject@gmail.com,
FB:www.facebook.com/embeddedplusproject
This document discusses different techniques for improving the quality of digitized signals, including companding, DPCM, and ADPCM. Companding uses non-uniform quantization to boost weak signals before quantization. DPCM transmits differences between actual and predicted sample values, reducing quantization noise. ADPCM further improves on DPCM with an adaptive quantizer that adjusts quantization step size based on prediction error. These methods allow higher quality digitization using fewer bits compared to basic uniform quantization.
Voice Activity Detection using Single Frequency FilteringTejus Adiga M
1. Voice Activity Detection (VAD) aims to locate speech segments in an input signal corrupted by noise by classifying frames as speech or noise.
2. Several time domain and frequency domain algorithms are discussed for VAD, including short-term energy, zero crossing rate, frequency subband distance measure, and long-term spectral flatness measure.
3. Single frequency filtering is also described, which analyzes envelopes at discrete frequencies to classify frames based on a learned noise floor.
The document introduces several MPEG standards for audio and video compression and transmission, including MPEG-1, MPEG-2, MPEG-4, MPEG-7, and MPEG-21. MPEG-1 was the first standard and defines lossy compression for video and audio. It became the most widely compatible format for these media. MPEG-2 built upon MPEG-1 and is used widely for digital television and DVD formats. MPEG-4 added new features like 3D rendering and interactivity. MPEG-7 defined standards for multimedia content description and MPEG-21 aims to define an open framework for multimedia applications and digital rights management.
The document provides information about MPEG compression standards. It discusses the history of MPEG and how it was established in 1988 as a joint effort between ISO and IEC to set standards for audio and video compression. It describes several MPEG standards including MPEG-1, MPEG-2, MPEG-4, MPEG-7, and MPEG-21. MPEG-4 is discussed in more detail, explaining that it offers greater efficiency than MPEG-2, allows encoding of mixed data types, and enables interaction of audio-visual scenes at the receiver end. The document contains diagrams and tables to illustrate key points about the different MPEG standards and compression techniques.
This document summarizes MPEG 1 and 2 video compression standards. It explains the need for video compression due to the large data rates of uncompressed video like HDTV. MPEG compression works by predicting frames from previous frames using motion compensation and coding the residual errors. It uses I, P, and B frames along with other techniques like chroma subsampling to achieve high compression ratios like 83:1 while maintaining quality. MPEG-2 improved upon MPEG-1 by supporting higher resolutions and bitrates needed for digital television.
The document discusses MPEG-2 video compression. It explains that MPEG-2 builds on MPEG-1 by providing backward compatibility and exploiting both intraframe and interframe redundancies to achieve high compression ratios. It describes how video frames are organized into Groups of Pictures (GOPs) containing I, P, and B frames. The compression steps of discrete cosine transform, weighting, re-quantization, entropy coding, and run length coding are explained. It also discusses how motion compensation of P and B frames further reduces file sizes by only encoding differences between frames.
Digital image processing involves compressing images to reduce file sizes. Image compression removes redundant data using three main techniques: coding redundancy reduction assigns shorter codes to more common pixel values; spatial and temporal redundancy reduction exploits correlations between neighboring pixel values; and irrelevant information removal discards visually unimportant data. Compression is achieved by an encoder that applies these techniques, while a decoder reconstructs the image for viewing. Popular compression methods include Huffman coding and arithmetic coding. Compression allows storage and transmission of images and video using less data while maintaining acceptable visual quality.
The document discusses image compression techniques. It outlines the fundamentals of image compression including encoding, decoding, and algorithms like JPEG and JPEG 2000. Image compression aims to reduce data storage and transmission requirements by removing redundant pixel information. It allows for more efficient sharing and storage of images across industries like printing, data storage, telecommunications, satellite imaging, and television.
This document provides an overview and illustration of JPEG 2000, a new image compression standard that replaces JPEG. It explains key features like tile division, progression order, quantization, discrete wavelet transformation, DC level shifting, and region of interest encoding. Graphs show that JPEG 2000 provides much smaller file sizes than JPEG while maintaining higher image quality. The conclusion states that JPEG 2000 offers excellent compression, fully exploits discrete wavelet transformation, and is well-suited for hardware implementation, establishing it as the most advanced image compression standard.
I will start with a question "why signal can be compressed?" I will then describe quantization, entropy-coding, difference-PCM, and Discrete Cosine Transform (DCT). My main motive will be to illustrate the basic principle rather than to describe the details of each method. Finally I will discuss how these various algorithm combined to get the JPEG standard for image compression. Time permitting, I will comment on various famous theories which lead to JPEG standard.
From the Un-Distinguished Lecture Series (http://ws.cs.ubc.ca/~udls/). The talk was given May 18, 2007.
The document discusses audio compression techniques. It begins with an introduction to pulse code modulation (PCM) and then describes μ-law and A-law compression standards which compress audio using companding algorithms. It also covers differential PCM and adaptive differential PCM (ADPCM) techniques. The document then discusses the MPEG audio compression standard, including its encoder architecture, three layer standards (Layers I, II, III), and applications. It concludes with a comparison of various MPEG audio compression standards and references.
This document summarizes a communications lab on signal statistics and an introduction to Simulink. The first part discusses calculating statistics of signals such as maximum/minimum values, mean, energy, and root mean squared value to quantify signal quality. It provides examples of analyzing a clarinet sound signal. The second part introduces Simulink for dynamic signal simulation. It describes using Simulink to generate a sinusoidal signal, view it on an oscilloscope, and analyze its power spectral density with a spectrum analyzer block. The third part discusses building a simple Simulink model with these components to familiarize with the basics of the tool.
Task allocation and scheduling inmultiprocessorsDon William
This document discusses task allocation and scheduling in a multi-processor environment. It describes generating synthetic tasks and assigning them priorities using static and dynamic scheduling algorithms like Rate Monotonic and Earliest Deadline First. It then covers allocating tasks to processors using algorithms like Next Fit and Bin Packing to optimize processor utilization. The goal is to schedule tasks dynamically in a multi-processor system to improve performance over uniprocessor scheduling.
The document discusses different types of multimedia compression formats including GIF, JPEG, MPEG, and WAV. It explains the differences between lossy compression formats like JPEG and MPEG which can lose some data quality versus lossless formats like GIF and ZIP. The document also covers color spaces, digital image representation, and the compression algorithms used in common formats such as GIF using LZW compression and JPEG using discrete cosine transform quantization.
This document provides an overview of MPEG-1 audio compression. It describes the key components of the MPEG-1 audio encoder including the polyphase filter bank that transforms audio into frequency subbands, the psychoacoustic model that determines inaudible parts of the signal, and the coding and bit allocation process that assigns bits to subbands. The overview concludes by noting that MPEG-1 audio provides high compression while retaining quality and paved the way for future audio compression standards.
Sparse Approximation of Gram Matrices for GMMN-based Speech SynthesisTomoki Koriyama
The document examines approximation methods for reducing the computational cost of training generative moment matching network (GMMN)-based speech synthesis. It investigates approximating the Gram matrices using block diagonal and random Fourier feature approaches, as well as using random and clustering-based methods for minibatch selection. Subjective tests found that the random Fourier feature and clustering-based minibatch selection approaches achieved higher subjective scores for inter-utterance variation while maintaining naturalness.
This document summarizes video compression techniques used in standards like MPEG. It discusses how video contains redundancies that compression reduces by encoding differences between frames and subsampling color. The MPEG algorithm compresses video in 5 steps: resolution reduction, motion estimation between frame types (I, P, B frames), discrete cosine transform, quantization, and entropy encoding. Standards like MPEG-1, 2, and 4 provide different compression ratios and capabilities for various applications like streaming video.
The document discusses video compression basics and MPEG-2 video compression. It explains that video frames contain redundant spatial and temporal data that can be compressed. MPEG-2 uses three frame types (I, P, B frames) and compresses frames using intra-frame and inter-frame encoding techniques like DCT, quantization, and entropy encoding to remove redundancy. The encoding process transforms raw video frames to compressed bitstreams for efficient storage and transmission.
Signal and image processing on satellite communication using MATLABEmbedded Plus Trichy
Basic Explanations about satellite imaging and signal processing with the help of MATLAB.
Contact us: 23,Nandhi koil Street, Near Nakoda Showroom,Theppakulam,Trichy
Mb.No:9360212155.
Mail:embeddedplusproject@gmail.com,
FB:www.facebook.com/embeddedplusproject
This document discusses different techniques for improving the quality of digitized signals, including companding, DPCM, and ADPCM. Companding uses non-uniform quantization to boost weak signals before quantization. DPCM transmits differences between actual and predicted sample values, reducing quantization noise. ADPCM further improves on DPCM with an adaptive quantizer that adjusts quantization step size based on prediction error. These methods allow higher quality digitization using fewer bits compared to basic uniform quantization.
Voice Activity Detection using Single Frequency FilteringTejus Adiga M
1. Voice Activity Detection (VAD) aims to locate speech segments in an input signal corrupted by noise by classifying frames as speech or noise.
2. Several time domain and frequency domain algorithms are discussed for VAD, including short-term energy, zero crossing rate, frequency subband distance measure, and long-term spectral flatness measure.
3. Single frequency filtering is also described, which analyzes envelopes at discrete frequencies to classify frames based on a learned noise floor.
The document introduces several MPEG standards for audio and video compression and transmission, including MPEG-1, MPEG-2, MPEG-4, MPEG-7, and MPEG-21. MPEG-1 was the first standard and defines lossy compression for video and audio. It became the most widely compatible format for these media. MPEG-2 built upon MPEG-1 and is used widely for digital television and DVD formats. MPEG-4 added new features like 3D rendering and interactivity. MPEG-7 defined standards for multimedia content description and MPEG-21 aims to define an open framework for multimedia applications and digital rights management.
The document provides information about MPEG compression standards. It discusses the history of MPEG and how it was established in 1988 as a joint effort between ISO and IEC to set standards for audio and video compression. It describes several MPEG standards including MPEG-1, MPEG-2, MPEG-4, MPEG-7, and MPEG-21. MPEG-4 is discussed in more detail, explaining that it offers greater efficiency than MPEG-2, allows encoding of mixed data types, and enables interaction of audio-visual scenes at the receiver end. The document contains diagrams and tables to illustrate key points about the different MPEG standards and compression techniques.
This document summarizes MPEG 1 and 2 video compression standards. It explains the need for video compression due to the large data rates of uncompressed video like HDTV. MPEG compression works by predicting frames from previous frames using motion compensation and coding the residual errors. It uses I, P, and B frames along with other techniques like chroma subsampling to achieve high compression ratios like 83:1 while maintaining quality. MPEG-2 improved upon MPEG-1 by supporting higher resolutions and bitrates needed for digital television.
The document discusses MPEG-2 video compression. It explains that MPEG-2 builds on MPEG-1 by providing backward compatibility and exploiting both intraframe and interframe redundancies to achieve high compression ratios. It describes how video frames are organized into Groups of Pictures (GOPs) containing I, P, and B frames. The compression steps of discrete cosine transform, weighting, re-quantization, entropy coding, and run length coding are explained. It also discusses how motion compensation of P and B frames further reduces file sizes by only encoding differences between frames.
Digital image processing involves compressing images to reduce file sizes. Image compression removes redundant data using three main techniques: coding redundancy reduction assigns shorter codes to more common pixel values; spatial and temporal redundancy reduction exploits correlations between neighboring pixel values; and irrelevant information removal discards visually unimportant data. Compression is achieved by an encoder that applies these techniques, while a decoder reconstructs the image for viewing. Popular compression methods include Huffman coding and arithmetic coding. Compression allows storage and transmission of images and video using less data while maintaining acceptable visual quality.
The document discusses image compression techniques. It outlines the fundamentals of image compression including encoding, decoding, and algorithms like JPEG and JPEG 2000. Image compression aims to reduce data storage and transmission requirements by removing redundant pixel information. It allows for more efficient sharing and storage of images across industries like printing, data storage, telecommunications, satellite imaging, and television.
This document provides an overview and illustration of JPEG 2000, a new image compression standard that replaces JPEG. It explains key features like tile division, progression order, quantization, discrete wavelet transformation, DC level shifting, and region of interest encoding. Graphs show that JPEG 2000 provides much smaller file sizes than JPEG while maintaining higher image quality. The conclusion states that JPEG 2000 offers excellent compression, fully exploits discrete wavelet transformation, and is well-suited for hardware implementation, establishing it as the most advanced image compression standard.
I will start with a question "why signal can be compressed?" I will then describe quantization, entropy-coding, difference-PCM, and Discrete Cosine Transform (DCT). My main motive will be to illustrate the basic principle rather than to describe the details of each method. Finally I will discuss how these various algorithm combined to get the JPEG standard for image compression. Time permitting, I will comment on various famous theories which lead to JPEG standard.
From the Un-Distinguished Lecture Series (http://ws.cs.ubc.ca/~udls/). The talk was given May 18, 2007.
The document discusses audio compression techniques. It begins with an introduction to pulse code modulation (PCM) and then describes μ-law and A-law compression standards which compress audio using companding algorithms. It also covers differential PCM and adaptive differential PCM (ADPCM) techniques. The document then discusses the MPEG audio compression standard, including its encoder architecture, three layer standards (Layers I, II, III), and applications. It concludes with a comparison of various MPEG audio compression standards and references.
This document summarizes a communications lab on signal statistics and an introduction to Simulink. The first part discusses calculating statistics of signals such as maximum/minimum values, mean, energy, and root mean squared value to quantify signal quality. It provides examples of analyzing a clarinet sound signal. The second part introduces Simulink for dynamic signal simulation. It describes using Simulink to generate a sinusoidal signal, view it on an oscilloscope, and analyze its power spectral density with a spectrum analyzer block. The third part discusses building a simple Simulink model with these components to familiarize with the basics of the tool.
Task allocation and scheduling inmultiprocessorsDon William
This document discusses task allocation and scheduling in a multi-processor environment. It describes generating synthetic tasks and assigning them priorities using static and dynamic scheduling algorithms like Rate Monotonic and Earliest Deadline First. It then covers allocating tasks to processors using algorithms like Next Fit and Bin Packing to optimize processor utilization. The goal is to schedule tasks dynamically in a multi-processor system to improve performance over uniprocessor scheduling.
The document discusses different types of multimedia compression formats including GIF, JPEG, MPEG, and WAV. It explains the differences between lossy compression formats like JPEG and MPEG which can lose some data quality versus lossless formats like GIF and ZIP. The document also covers color spaces, digital image representation, and the compression algorithms used in common formats such as GIF using LZW compression and JPEG using discrete cosine transform quantization.
This document provides an overview of MPEG-1 audio compression. It describes the key components of the MPEG-1 audio encoder including the polyphase filter bank that transforms audio into frequency subbands, the psychoacoustic model that determines inaudible parts of the signal, and the coding and bit allocation process that assigns bits to subbands. The overview concludes by noting that MPEG-1 audio provides high compression while retaining quality and paved the way for future audio compression standards.
DataScience Lab 2017_Кто здесь? Автоматическая разметка спикеров на телефонны...GeeksLab Odessa
DataScience Lab, 13 мая 2017
Кто здесь? Автоматическая разметка спикеров на телефонных разговорах
Юрий Гуц (Machine Learning Engineer, DataRobot)
Автоматическая аннотация спикеров — интересная задача в обработке мультимедиа-данных. Нам нужно ответить на вопрос "Кто говорит когда?", не зная ничего о количестве и личности спикеров, присутствующих на записи. В этом докладе мы рассмотрим работающие методы для аннотации спикеров на телефонных разговорах.
Все материалы: http://datascience.in.ua/report2017
The document summarizes a real-time automatic speaker segmentation system. The system uses an audio analysis front-end to extract features from short audio frames and classify them as voiced, unvoiced or silence. Speaker models are incrementally updated using a quasi-GMM approach on only voiced frames. Speaker changes are detected using Bayesian Information Criterion on the speaker models, and validated to reduce false alarms. The system was tested on broadcast news and telephone conversation datasets, achieving real-time operation with tuned parameters.
This document discusses using Gamma tone Frequency Cepstral Coefficients (GFCC) and K-means clustering to identify singers based on their voice. It begins by explaining that MFCC is not accurate in noisy environments, while GFCC performs well in both clean and noisy audio. The process involves extracting GFCC features from the audio, using K-means clustering to group similar voices into clusters, and dynamic time warping for authentication. Feature extraction with GFCC involves preprocessing, framing, windowing, computing the discrete Fourier transform, applying a gamma tone filter bank, logarithmic compression, and discrete cosine transformation to generate feature vectors. K-means clustering is then used to group the feature vectors from similar voices into clusters to identify
The document discusses audio quantization and transmission. It covers:
1) Quantization converts continuous audio signals into discrete digital signals by sampling and assigning numeric codes, which are then transmitted or stored.
2) Compression uses linear or non-linear quantization, with non-linear providing better protection of quiet passages.
3) Common compression techniques include pulse code modulation (PCM), differential PCM (DPCM), adaptive DPCM (ADPCM) which adapt the quantizer and predictor to the audio signal.
This document presents the Improved Cepstra Minimum-Mean-Square-Error (ICMMSE) noise reduction algorithm for robust speech recognition. ICMMSE improves on the previous CMMSE algorithm in several ways: it uses an improved minimum controlled recursive averaging algorithm to estimate speech probability more accurately, refines prior signal-to-noise ratio estimation, applies gain smoothing or optimally-modified log-spectral amplitude processing to modify the gain function, and performs two-stage noise reduction processing. Experiments on the Aurora 2, CHiME-3, and Cortana tasks show ICMMSE consistently outperforms CMMSE and baseline systems, achieving relative word error rate reductions of up to 25%.
The document summarizes techniques for text-independent speaker recognition from audio signals. It discusses the principles of automatic speaker recognition including identification and verification. The key steps are voice recording, feature extraction using MFCC, modeling reference models for each speaker, pattern matching of input features against models, and making an identification or verification decision. Feature extraction involves framing the audio, windowing, FFT, mapping frequencies to the mel scale, and taking the DCT to produce cepstral coefficients.
Text independent speaker recognition systemDeepesh Lekhak
This document outlines a project to develop a text-independent speaker recognition system. It lists the project members and provides an overview of the presentation sections, which include the system architecture, methodology, results and analysis, and applications. The methodology section describes implementing the system in MATLAB, including voice capturing, pre-processing, MFCC feature extraction, GMM matching, and identification/verification. It also outlines implementing the system on an FPGA, including analog conversion, storage, framing, FFT, mel spectrum, MFCC extraction, and UART transmission to MATLAB for further processing. The results show over 99% recognition accuracy with longer training and test data.
Weakly-Supervised Sound Event Detection with Self-AttentionNU_I_TODALAB
IEEE ICASSP 2020
Koichi Miyazaki, Tatsuya Komatsu, Tomoki Hayashi, Shinji Watanabe, Tomoki Toda, Kazuya Takeda, Weakly-supervised sound event detection with self-attention, May 2020
Toda Laboratory, Department of Intelligent Systems, Graduate School of Informatics, Nagoya University
The following resources come from the 2009/10 B.Sc in Media Technology and Digital Broadcast (course number 2ELE0076) from the University of Hertfordshire. All the mini projects are designed as level two modules of the undergraduate programmes.
Generating a time shrunk lecture video by eventYara Ali
The document describes a system for generating time-shortened lecture videos through event detection. The system records high-resolution lecture footage and then analyzes it to detect speech periods using voice activity detection and chalkboard writing periods using object detection and segmentation techniques. These event detection techniques allow the system to automatically generate a shortened lecture video by removing non-event periods and fast-forwarding writing periods. Evaluation of the system on test lecture recordings showed it could generate videos 20-30% shorter than the originals, similar to manual shortening by humans.
General Kalman Filter & Speech Enhancement for Speaker Identificationijcisjournal
Presence of noise increases the dimension of the information. A noise suppression algorithm is developed
with an idea of combining the General Kalman Filter and Estimate Maximization (EM) frame work.This
combination is helpful and effective in identifying noise characteristics of an acoustic environment.
Recursion between Estimate step and Maximization step enabled the algorithm to deal any model of noise.
The same Speech enhancement procedure in applied in the pre-processing stage of a conventional Speaker
identification method. Due to the non-stationary nature of noise and speech adaptive algorithms are
required. Algorithm is first applied for Speech enhancement problem and then extended to using it in the
pre-processing step of the Speaker identification. The present work is compared in terms of significant
metrics with existing and popular algorithms and results show that the developed algorithm is dominant
over them.
Ukrainian Catholic University
Faculty of Applied Sciences
Data Science Master Program
January 23rd
Abstract. Speaker classification is an essential task in the machine learning domain, with many practical applications in identification and natural language processing. This work concentrates on speaker classification as a subtask of general speaker diarization for real-world conversation scenarios. We research the domain of modern speech processing and present the original speaker classification approach based on the recent developments in convolutional neural networks. Our method uses a spectrogram as input to the CNN classifier model, allowing it to capture spatial information about voice frequencies distribution. Presented results show beyond human ability performance and give strong prospects for future development.
Audio tagging system using densely connected convolutional networks (DCASE201...Hyun-gui Lim
In this talk, we will explain the techniques and models applied to the submission of Cochlear.ai team for DCASE 2018 task 2: General-purpose audio tagging of Freesound content with AudioSet labels. We mainly focused on how to train deep learning models efficiently against strong augmentation and label noise. First, we conducted a single-block DenseNet architecture and multi-head softmax classifier for efficient learning with mixup augmentation. For the label noise, we applied the batch-wise loss masking to eliminate the loss of outliers in a mini-batch. We also tried an ensemble of various models, trained by using different sampling rate or audio representation.
The document summarizes an academic paper on audio inpainting using generative adversarial networks (GANs). It discusses using a dual discriminator Wasserstein GAN (D2WGAN) model to fill in missing audio content of 500-550ms by extracting short and long-range borders. The model was tested on piano, guitar, and piano+orchestra datasets, with the guitar dataset achieving the best results. Increasing the training steps significantly improved performance without overfitting. Future work could explore different border lengths, additional network layers, and combining waveforms and spectrograms.
Sara Hooker & Sean McPherson, Delta Analytics, at MLconf Seattle 2017MLconf
This document provides information about Delta Analytics, a non-profit organization that provides pro bono data consulting services to social sector organizations. It discusses Delta Analytics' work with Rainforest Connection, including developing machine learning models to detect chainsaw sounds from audio data collected by recycled cell phones deployed in rainforests. Key points discussed include developing convolutional neural networks to classify audio spectrograms, addressing challenges like limited labelled training data and unknown guardian positions, and experiments to estimate the direction of detected sounds.
Real-Time Non-Intrusive Speech Quality Estimation: A Signal-Based Modeladil raja
The document presents a non-intrusive speech quality estimation model developed using genetic programming. It discusses existing subjective and objective speech quality assessment methods. The proposed model uses genetic programming to derive a mapping from features extracted from speech signals using ITU-T P.563 to estimated mean opinion scores. The model outperforms the reference P.563 method with a 9.89% reduction in training error and 16.41% reduction in testing error. Key features selected by the model relate to vocal tract characteristics, distortion levels, and spectral properties.
Conv-TasNet is a convolutional neural network architecture for speech separation that directly operates on the raw audio waveform in the time domain. It surpasses previous approaches that perform separation in the time-frequency domain using magnitude masking. Conv-TasNet uses a convolutional encoder to transform short audio segments into representations, estimates separation masks for each source, and uses a convolutional decoder to reconstruct the sources. Experiments show Conv-TasNet achieves substantially better speech separation than ideal time-frequency masking baselines as measured by metrics like SI-SNRi and SDRi. The best performing model uses a small filter length, increasing numbers of encoder/decoder filters, and deep convolutional layers.
Similar to Dcase2016 oral presentation - Experiments on DCASE 2016: Acoustic Scene Classification and Sound Event Detection in real life recordings (20)
Fueling AI with Great Data with Airbyte WebinarZilliz
This talk will focus on how to collect data from a variety of sources, leveraging this data for RAG and other GenAI use cases, and finally charting your course to productionalization.
HCL Notes and Domino License Cost Reduction in the World of DLAUpanagenda
Webinar Recording: https://www.panagenda.com/webinars/hcl-notes-and-domino-license-cost-reduction-in-the-world-of-dlau/
The introduction of DLAU and the CCB & CCX licensing model caused quite a stir in the HCL community. As a Notes and Domino customer, you may have faced challenges with unexpected user counts and license costs. You probably have questions on how this new licensing approach works and how to benefit from it. Most importantly, you likely have budget constraints and want to save money where possible. Don’t worry, we can help with all of this!
We’ll show you how to fix common misconfigurations that cause higher-than-expected user counts, and how to identify accounts which you can deactivate to save money. There are also frequent patterns that can cause unnecessary cost, like using a person document instead of a mail-in for shared mailboxes. We’ll provide examples and solutions for those as well. And naturally we’ll explain the new licensing model.
Join HCL Ambassador Marc Thomas in this webinar with a special guest appearance from Franz Walder. It will give you the tools and know-how to stay on top of what is going on with Domino licensing. You will be able lower your cost through an optimized configuration and keep it low going forward.
These topics will be covered
- Reducing license cost by finding and fixing misconfigurations and superfluous accounts
- How do CCB and CCX licenses really work?
- Understanding the DLAU tool and how to best utilize it
- Tips for common problem areas, like team mailboxes, functional/test users, etc
- Practical examples and best practices to implement right away
Generating privacy-protected synthetic data using Secludy and MilvusZilliz
During this demo, the founders of Secludy will demonstrate how their system utilizes Milvus to store and manipulate embeddings for generating privacy-protected synthetic data. Their approach not only maintains the confidentiality of the original data but also enhances the utility and scalability of LLMs under privacy constraints. Attendees, including machine learning engineers, data scientists, and data managers, will witness first-hand how Secludy's integration with Milvus empowers organizations to harness the power of LLMs securely and efficiently.
HCL Notes und Domino Lizenzkostenreduzierung in der Welt von DLAUpanagenda
Webinar Recording: https://www.panagenda.com/webinars/hcl-notes-und-domino-lizenzkostenreduzierung-in-der-welt-von-dlau/
DLAU und die Lizenzen nach dem CCB- und CCX-Modell sind für viele in der HCL-Community seit letztem Jahr ein heißes Thema. Als Notes- oder Domino-Kunde haben Sie vielleicht mit unerwartet hohen Benutzerzahlen und Lizenzgebühren zu kämpfen. Sie fragen sich vielleicht, wie diese neue Art der Lizenzierung funktioniert und welchen Nutzen sie Ihnen bringt. Vor allem wollen Sie sicherlich Ihr Budget einhalten und Kosten sparen, wo immer möglich. Das verstehen wir und wir möchten Ihnen dabei helfen!
Wir erklären Ihnen, wie Sie häufige Konfigurationsprobleme lösen können, die dazu führen können, dass mehr Benutzer gezählt werden als nötig, und wie Sie überflüssige oder ungenutzte Konten identifizieren und entfernen können, um Geld zu sparen. Es gibt auch einige Ansätze, die zu unnötigen Ausgaben führen können, z. B. wenn ein Personendokument anstelle eines Mail-Ins für geteilte Mailboxen verwendet wird. Wir zeigen Ihnen solche Fälle und deren Lösungen. Und natürlich erklären wir Ihnen das neue Lizenzmodell.
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.
Diese Themen werden behandelt
- Reduzierung der Lizenzkosten durch Auffinden und Beheben von Fehlkonfigurationen und überflüssigen Konten
- Wie funktionieren CCB- und CCX-Lizenzen wirklich?
- Verstehen des DLAU-Tools und wie man es am besten nutzt
- Tipps für häufige Problembereiche, wie z. B. Team-Postfächer, Funktions-/Testbenutzer usw.
- Praxisbeispiele und Best Practices zum sofortigen Umsetzen
Trusted Execution Environment for Decentralized Process MiningLucaBarbaro3
Presentation of the paper "Trusted Execution Environment for Decentralized Process Mining" given during the CAiSE 2024 Conference in Cyprus on June 7, 2024.
GraphRAG for Life Science to increase LLM accuracyTomaz Bratanic
GraphRAG for life science domain, where you retriever information from biomedical knowledge graphs using LLMs to increase the accuracy and performance of generated answers
Introduction of Cybersecurity with OSS at Code Europe 2024Hiroshi SHIBATA
I develop the Ruby programming language, RubyGems, and Bundler, which are package managers for Ruby. Today, I will introduce how to enhance the security of your application using open-source software (OSS) examples from Ruby and RubyGems.
The first topic is CVE (Common Vulnerabilities and Exposures). I have published CVEs many times. But what exactly is a CVE? I'll provide a basic understanding of CVEs and explain how to detect and handle vulnerabilities in OSS.
Next, let's discuss package managers. Package managers play a critical role in the OSS ecosystem. I'll explain how to manage library dependencies in your application.
I'll share insights into how the Ruby and RubyGems core team works to keep our ecosystem safe. By the end of this talk, you'll have a better understanding of how to safeguard your code.
leewayhertz.com-AI in predictive maintenance Use cases technologies benefits ...alexjohnson7307
Predictive maintenance is a proactive approach that anticipates equipment failures before they happen. At the forefront of this innovative strategy is Artificial Intelligence (AI), which brings unprecedented precision and efficiency. AI in predictive maintenance is transforming industries by reducing downtime, minimizing costs, and enhancing productivity.
Building Production Ready Search Pipelines with Spark and MilvusZilliz
Spark is the widely used ETL tool for processing, indexing and ingesting data to serving stack for search. Milvus is the production-ready open-source vector database. In this talk we will show how to use Spark to process unstructured data to extract vector representations, and push the vectors to Milvus vector database for search serving.
Digital Marketing Trends in 2024 | Guide for Staying AheadWask
https://www.wask.co/ebooks/digital-marketing-trends-in-2024
Feeling lost in the digital marketing whirlwind of 2024? Technology is changing, consumer habits are evolving, and staying ahead of the curve feels like a never-ending pursuit. This e-book is your compass. Dive into actionable insights to handle the complexities of modern marketing. From hyper-personalization to the power of user-generated content, learn how to build long-term relationships with your audience and unlock the secrets to success in the ever-shifting digital landscape.
Ocean lotus Threat actors project by John Sitima 2024 (1).pptxSitimaJohn
Ocean Lotus cyber threat actors represent a sophisticated, persistent, and politically motivated group that poses a significant risk to organizations and individuals in the Southeast Asian region. Their continuous evolution and adaptability underscore the need for robust cybersecurity measures and international cooperation to identify and mitigate the threats posed by such advanced persistent threat groups.
TrustArc Webinar - 2024 Global Privacy SurveyTrustArc
How does your privacy program stack up against your peers? What challenges are privacy teams tackling and prioritizing in 2024?
In the fifth annual Global Privacy Benchmarks Survey, we asked over 1,800 global privacy professionals and business executives to share their perspectives on the current state of privacy inside and outside of their organizations. This year’s report focused on emerging areas of importance for privacy and compliance professionals, including considerations and implications of Artificial Intelligence (AI) technologies, building brand trust, and different approaches for achieving higher privacy competence scores.
See how organizational priorities and strategic approaches to data security and privacy are evolving around the globe.
This webinar will review:
- The top 10 privacy insights from the fifth annual Global Privacy Benchmarks Survey
- The top challenges for privacy leaders, practitioners, and organizations in 2024
- Key themes to consider in developing and maintaining your privacy program
Main news related to the CCS TSI 2023 (2023/1695)Jakub Marek
An English 🇬🇧 translation of a presentation to the speech I gave about the main changes brought by CCS TSI 2023 at the biggest Czech conference on Communications and signalling systems on Railways, which was held in Clarion Hotel Olomouc from 7th to 9th November 2023 (konferenceszt.cz). Attended by around 500 participants and 200 on-line followers.
The original Czech 🇨🇿 version of the presentation can be found here: https://www.slideshare.net/slideshow/hlavni-novinky-souvisejici-s-ccs-tsi-2023-2023-1695/269688092 .
The videorecording (in Czech) from the presentation is available here: https://youtu.be/WzjJWm4IyPk?si=SImb06tuXGb30BEH .
Monitoring and Managing Anomaly Detection on OpenShift.pdfTosin Akinosho
Monitoring and Managing Anomaly Detection on OpenShift
Overview
Dive into the world of anomaly detection on edge devices with our comprehensive hands-on tutorial. This SlideShare presentation will guide you through the entire process, from data collection and model training to edge deployment and real-time monitoring. Perfect for those looking to implement robust anomaly detection systems on resource-constrained IoT/edge devices.
Key Topics Covered
1. Introduction to Anomaly Detection
- Understand the fundamentals of anomaly detection and its importance in identifying unusual behavior or failures in systems.
2. Understanding Edge (IoT)
- Learn about edge computing and IoT, and how they enable real-time data processing and decision-making at the source.
3. What is ArgoCD?
- Discover ArgoCD, a declarative, GitOps continuous delivery tool for Kubernetes, and its role in deploying applications on edge devices.
4. Deployment Using ArgoCD for Edge Devices
- Step-by-step guide on deploying anomaly detection models on edge devices using ArgoCD.
5. Introduction to Apache Kafka and S3
- Explore Apache Kafka for real-time data streaming and Amazon S3 for scalable storage solutions.
6. Viewing Kafka Messages in the Data Lake
- Learn how to view and analyze Kafka messages stored in a data lake for better insights.
7. What is Prometheus?
- Get to know Prometheus, an open-source monitoring and alerting toolkit, and its application in monitoring edge devices.
8. Monitoring Application Metrics with Prometheus
- Detailed instructions on setting up Prometheus to monitor the performance and health of your anomaly detection system.
9. What is Camel K?
- Introduction to Camel K, a lightweight integration framework built on Apache Camel, designed for Kubernetes.
10. Configuring Camel K Integrations for Data Pipelines
- Learn how to configure Camel K for seamless data pipeline integrations in your anomaly detection workflow.
11. What is a Jupyter Notebook?
- Overview of Jupyter Notebooks, an open-source web application for creating and sharing documents with live code, equations, visualizations, and narrative text.
12. Jupyter Notebooks with Code Examples
- Hands-on examples and code snippets in Jupyter Notebooks to help you implement and test anomaly detection models.
3. Overview
1. Task 1: Acoustic Scene Classification
a. Approach
b. Results on challenge
c. Conclusions
2. Task 3: Sound Event Detection in Real Life Recordings
a. Approach
b. Results on challenge
c. Conclusions
8. Approach
Compute
MFCC
Train
UBM
Compute
high-level
features
● The high-level features try to capture the distribution of the scene’s MFCCs.
● Compute features, labeled as Beta, by adapting the UBM to the MFCCs using
MAP* for each component size.
○ Four supervectors with stacked means.
○ Four supervectors with stacked means and variance.
9. Approach
Compute
MFCC
Train
UBM
Compute
high-level
features
● The high-level features try to capture the distribution of the scene’s MFCCs.
● Compute features, labeled as Beta, by adapting the UBM to the MFCCs using
MAP* for each component size.
○ Four supervectors with stacked means.
○ Four supervectors with stacked means and variance.
MAP*
10. Approach
● Train SVM with Linear and RBF kernels for each of the 8 supervectors, thus 16
systems in total.
Train/Classify
SVM
Compute
MFCC
Train
UBM
Compute
high-level
features
11. Approach
● Fuse classifiers based on maximum prediction vote.
Fuse
classifiers
Output
Scores
Train/Classify
SVM
Compute
MFCC
Train
UBM
Compute
high-level
features
16. Conclusions
● GMM-based representations capture well the content of long scene recordings.
○ For example, i-vectors which got the best performance.
● Fusion of different beta vectors and kernels improved performance.
● We tried Alpha features which are histograms, similar to Bag-of-Audio-Words
representations in combination with 7 kernels, but didn’t help.
17. Task 3: Sound Event Detection in
Real Life Recordings
Approach
19. Approach: Standard
● Create a pipeline for each scene: Home and Residential Area.
● For training: extract the sound events from the scene recordings to have
isolated files.
Extract
sounds
Train set
20. Approach: Standard
● Merge both audio channels into one..
● Compute MFCCs with 13 dim. +D+DD, 30 ms window and 50% overlap.
○ Also explored: Separable Gabor Filter Banks, Scatnet and Stacked with and without PCA.
Extract
sounds
Compute
MFCC
21. Approach: Standard
● Tpot is a toolbox that automatically creates machine learning pipelines.
● Tpot (v4) tests and optimizes 12 classifiers such as SVMs, Decision Trees,
K-Neighbors, Logistic Regression, etc.
● Main selections: Random Forest, Logistic Regression, Gradient Boosting
Extract
sounds
Compute
MFCC
Generate
TPOT
pipeline
Train
Classifier
23. Approach: Generic
● The generic class was created to address out-of-vocabulary sound events.
● The class is specific to each scene.
● The number of audio files corresponds to the average of files per class.
Extract
sounds
Compute
MFCC
Segment
audio
Output
Scores
Generic
Sound
Event
Classify
segments
24. Approach: Perturbation
● The time-perturbation is intended to address robustness and compensate for
small data.
● The signal was speeded up to 2x and slowed down to 0.3x to output 13
different versions of the original file.
Extract
sounds
Compute
MFCC
Segment
audio
Output
Scores
Time
perturbation
Classify
segments
25. Task 3: Sound Event Detection in
Real Life Recordings
Results
26. Results on challenge
Overall Segment-based Error Rate
Run Development Evaluation
Baseline 0.91 0.87
Best 0.91S
0.80S
27. Results on challenge
Overall Segment-based Error Rate
Run Development Evaluation
Standard (S) 0.84 1.05
Baseline 0.91 0.87
Best 0.91S
0.80S
28. Results on challenge
Overall Segment-based Error Rate
Run Development Evaluation
Standard (S) 0.84 1.05
S+Generic 0.81 1.07
Baseline 0.91 0.87
Best 0.91S
0.80S
29. Results on challenge
Overall Segment-based Error Rate
Run Development Evaluation
Standard (S) 0.84 1.05
S+Generic 0.81 1.07
S+Generic+Perturbation 0.76 0.961
Baseline 0.91 0.87
Best 0.91S
0.80S
9th
30. Results on challenge
Overall Segment-based Error Rate
Run Development Evaluation
Standard (S) 0.84 1.05
S+Generic 0.81 1.07
S+Generic+Perturbation 0.76 0.961
*S+Perturbation 0.74 0.96
Baseline 0.91 0.87
Best 0.91S
0.80S
9th
37. Per-scene results on the challenge
Scene Rank - ex aequo
Residential Area 1 tiers
Forest, Grocery Store, Metro, Office 2
Train, Home, City Center, Bus 3
Beach, Cafe, Car, Tram 4
Library, Park 5
38. Sound-events results on challenge
S+Generic+Perturbation
Home Residential Rank ...
Cupboard Wind Blowing 1
Glass Jingling Children Shouting 2
... ... ...
Water Tap Running People Speaking 9
Object Impact Bird Singing 13