Motorola presents in collaboration with CNEL Introduction  Motivation: The limitation of traditional narrowband transmission channel  Advantage: Phone.

Slides:



Advertisements
Similar presentations
The Fully Networked Car Geneva, 4-5 March Automotive Speech Enhancement of Today: Applications, Challenges and Solutions Tim Haulick Harman/Becker.
Advertisements

SHORT-TIME MULTICHANNEL NOISE CORRELATION MATRIX ESTIMATORS FOR ACOUSTIC SIGNALS By: Jonathan Blanchette and Martin Bouchard.
Chunyi Peng, Guobin Shen, Yongguang Zhang, Yanlin Li, Kun Tan BeepBeep: A High Accuracy Acoustic Ranging System using COTS Mobile Devices.
Speech Enhancement through Noise Reduction By Yating & Kundan.
1 Small-scale Mobile radio propagation Small-scale Mobile radio propagation l Small scale propagation implies signal quality in a short distance or time.
Improvement of Audio Capture in Handheld Devices through Digital Filtering Problem Microphones in handheld devices are of low quality to reduce cost. This.
Microphone Array Post-filter based on Spatially- Correlated Noise Measurements for Distant Speech Recognition Kenichi Kumatani, Disney Research, Pittsburgh.
G. Valenzise *, L. Gerosa, M. Tagliasacchi *, F. Antonacci *, A. Sarti * IEEE Int. Conf. On Advanced Video and Signal-based Surveillance, 2007 * Dipartimento.
Speech Coding Nicola Orio Dipartimento di Ingegneria dell’Informazione IV Scuola estiva AISV, 8-12 settembre 2008.
Blind Source Separation of Acoustic Signals Based on Multistage Independent Component Analysis Hiroshi SARUWATARI, Tsuyoki NISHIKAWA, and Kiyohiro SHIKANO.
1/44 1. ZAHRA NAGHSH JULY 2009 BEAM-FORMING 2/44 2.
Overview of Adaptive Multi-Rate Narrow Band (AMR-NB) Speech Codec
2003 MSS BA C-8 1 Acoustic Source Estimation with Doppler Processing Richard J. Kozick Bucknell University Brian M. Sadler Army Research Laboratory.
MPEG Audio Compression by V. Loumos. Introduction Motion Picture Experts Group (MPEG) International Standards Organization (ISO) First High Fidelity Audio.
3/24/2006Lecture notes for Speech Communications Multi-channel speech enhancement Chunjian Li DICOM, Aalborg University.
APPLICATION OF SPACE-TIME CODING TECHNIQUES IN THIRD GENERATION SYSTEMS - A. G. BURR ADAPTIVE SPACE-TIME SIGNAL PROCESSING AND CODING – A. G. BURR.
Audio Source Separation And ICA by Mike Davies & Nikolaos Mitianoudis Digital Signal Processing Lab Queen Mary, University of London.
1 New Technique for Improving Speech Intelligibility for the Hearing Impaired Miriam Furst-Yust School of Electrical Engineering Tel Aviv University.
HIWIRE meeting ITC-irst Activity report Marco Matassoni, Piergiorgio Svaizer March Torino.
So far: Historical overview of speech technology  basic components/goals for systems Quick review of DSP fundamentals Quick overview of pattern recognition.
Why is ASR Hard? Natural speech is continuous
Chunyi Peng, Guobin Shen, Yongguang Zhang, Yanlin Li, Kun Tan
A Full Frequency Masking Vocoder for Legal Eavesdropping Conversation Recording R. F. B. Sotero Filho, H. M. de Oliveira (qPGOM), R. Campello de Souza.
Zbigniew LEONOWICZ, Tadeusz LOBOS Wroclaw University of Technology Wroclaw University of Technology, Poland International Conference.
Introduction to Spectral Estimation
Normalization of the Speech Modulation Spectra for Robust Speech Recognition Xiong Xiao, Eng Siong Chng, and Haizhou Li Wen-Yi Chu Department of Computer.
Knowledge Base approach for spoken digit recognition Vijetha Periyavaram.
1 Robust HMM classification schemes for speaker recognition using integral decode Marie Roch Florida International University.
„Bandwidth Extension of Speech Signals“ 2nd Workshop on Wideband Speech Quality in Terminals and Networks: Assessment and Prediction 22nd and 23rd June.
ECE 8443 – Pattern Recognition ECE 8423 – Adaptive Signal Processing Objectives: Introduction SNR Gain Patterns Beam Steering Shading Resources: Wiki:
Two computations concerning fatigue damage and the Power Spectral Density Frank Sherratt.
EFFECTS OF MUTUAL COUPLING AND DIRECTIVITY ON DOA ESTIMATION USING MUSIC LOPAMUDRA KUNDU & ZHE ZHANG.
International Conference on Intelligent and Advanced Systems 2007 Chee-Ming Ting Sh-Hussain Salleh Tian-Swee Tan A. K. Ariff. Jain-De,Lee.
Reporter: Shih-Hsiang( 士翔 ). Introduction Speech signal carries information from many sources –Not all information is relevant or important for speech.
Blind speech dereverberation using multiple microphones Inseon JANG, Seungjin CHOI Intelligent Multimedia Lab Department of Computer Science and Engineering,
Multiuser Detection (MUD) Combined with array signal processing in current wireless communication environments Wed. 박사 3학기 구 정 회.
Compensating speaker-to-microphone playback system for robust speech recognition So-Young Jeong and Soo-Young Lee Brain Science Research Center and Department.
Ekapol Chuangsuwanich and James Glass MIT Computer Science and Artificial Intelligence Laboratory,Cambridge, Massachusetts 02139,USA 2012/07/2 汪逸婷.
Department of Electrical & Computer Engineering Auditory Perception Meena Ramani 04/09/2004.
Speech Signal Representations I Seminar Speech Recognition 2002 F.R. Verhage.
Communication Group Course Multidimensional DSP DoA Estimation Methods Pejman Taslimi – Spring 2009 Course Presentation – Amirkabir Univ Title: Acoustic.
Authors: Sriram Ganapathy, Samuel Thomas, and Hynek Hermansky Temporal envelope compensation for robust phoneme recognition using modulation spectrum.
Yi-zhang Cai, Jeih-weih Hung 2012/08/17 報告者:汪逸婷 1.
Estimation of Sound Source Direction Using Parabolic Reflection Board 2008 RISP International Workshop on Nonlinear Circuits and Signal Processing (NCSP’08)
Full-rank Gaussian modeling of convolutive audio mixtures applied to source separation Ngoc Q. K. Duong, Supervisor: R. Gribonval and E. Vincent METISS.
Voice Activity Detection based on OptimallyWeighted Combination of Multiple Features Yusuke Kida and Tatsuya Kawahara School of Informatics, Kyoto University,
Active Microphone with Parabolic Reflection Board for Estimation of Sound Source Direction Tetsuya Takiguchi, Ryoichi Takashima and Yasuo Ariki Organization.
Z bigniew Leonowicz, Wroclaw University of Technology Z bigniew Leonowicz, Wroclaw University of Technology, Poland XXIX  IC-SPETO.
Microphone Array Project ECE5525 – Speech Processing Robert Villmow 12/11/03.
Performance Comparison of Speaker and Emotion Recognition
Dr. Galal Nadim.  The root-MUltiple SIgnal Classification (root- MUSIC) super resolution algorithm is used for indoor channel characterization (estimate.
Professors: Eng. Diego Barral Eng. Mariano Llamedo Soria Julian Bruno
RCC-Mean Subtraction Robust Feature and Compare Various Feature based Methods for Robust Speech Recognition in presence of Telephone Noise Amin Fazel Sharif.
Analysis of Traction System Time-Varying Signals using ESPRIT Subspace Spectrum Estimation Method Z. Leonowicz, T. Lobos
Institut für Nachrichtengeräte und Datenverarbeitung Prof. Dr.-Ing. P. Vary On the Use of Artificial Bandwidth Extension Techniques in Wideband Speech.
Speech Enhancement using Excitation Source Information B. Yegnanarayana, S.R. Mahadeva Prasanna & K. Sreenivasa Rao Department of Computer Science & Engineering.
Benedikt Loesch and Bin Yang University of Stuttgart Chair of System Theory and Signal Processing International Workshop on Acoustic Echo and Noise Control,
Term paper on Smart antenna system
MOTOROLA Project 2004 Current members Prof. John G. Harris Dr. Mark D. Skowronski Meena Ramani, Ph.D. student Harsha Sathyendra, Ph.D. student Ismail Uysal,
ARENA08 Roma June 2008 Francesco Simeone (Francesco Simeone INFN Roma) Beam-forming and matched filter techniques.
UNIT-IV. Introduction Speech signal is generated from a system. Generation is via excitation of system. Speech travels through various media. Nature of.
Speech Enhancement Summer 2009
ARTIFICIAL NEURAL NETWORKS
Estimation Techniques for High Resolution and Multi-Dimensional Array Signal Processing EMS Group – Fh IIS and TU IL Electronic Measurements and Signal.
III. Analysis of Modulation Metrics IV. Modifications
Frequency Domain Perceptual Linear Predicton (FDPLP)
Submission Title: [Robust Ranging Algorithm for UWB radio]
Submission Title: [Robust Ranging Algorithm for UWB radio]
Submission Title: [Robust Ranging Algorithm for UWB radio]
Combination of Feature and Channel Compensation (1/2)
Presentation transcript:

Motorola presents in collaboration with CNEL Introduction  Motivation: The limitation of traditional narrowband transmission channel  Advantage: Phone line frequency range: 300Hz- 3400Hz; Recovered frequency range: 20Hz-8000Hz  Goal: Increase the speech intelligibility and quality by adding artificial high frequency components  Basic Assumption: The high correlation between the low-frequency and high-frequency components of the same phonemes  Frequency fold  GMM algorithm Meena Ramani, Lingyun Gu, Kausthub Kale Aim Method Captions to be set in Times or Times New Roman or equivalent, italic, 18 to 24 points, to the length of the column in case a figure takes more than 2/3 of column width. Bandwidth Expansion Direction of arrival estimation & Beamforming Speech enhancement for cell phones Use psychoacoustic and auditory system knowledge to improve speech loudness and intelligibility Motivation Need for enhanced voice quality Complete mobility under noisy conditions Ability to identify different speakers in a conference call Direction Of Arrival (DOA) estimation Real time operation Constraints Physical constraints Low software and hardware complexity Good performance at all frequencies Increase the intelligibility of speech Differentiate speech source from noise source Overcome problems of signal distortion due to noise Prevent loss of accuracy due to room reverberations DOA Algorithms Spatial Correlation methods Delay and SumMinimum Variance Subspace decomposition methods MUSIC Multiple Signal Estimation Coherent MUSIC Root MUSIC ESPRIT Estimation of Signal parameters using rotational invariance DOA Requirements DOA Algorithm requirements Low computational intensity (FLOPS) High accuracy (Confidence Interval) High speed (Time taken) Easy to implement Work well at low SNRs Work well in a 2 microphone narrow baseline (4cm) system. DOA Method Equation for Implementation Delay and Sum Minimum Variance MUSIC Coherent MUSIC Root MUSIC ESPRIT Low FLOPS count Good Accuracy High Speed and good low SNR performance LPC analysis Excitation regeneration WB LPC synthesis HPF Features extraction Spectral envelope estimation 1:2LPF + with and Hamming window length20 ms LPC order(wideband)18 LPC order(narrowband)14 Spectral representationLPC cepstrum Mixture number (Q)128 VQ codebook size128 Bandwidth Extension of Telephone Speech Excitation Regeneration Spectral Envelope Regeneration Project Golden Voice ® Golden voice Plot comparing the MSE for the six different methods at different SNRs Comparison of FLOPS for the six different methods for 10dB SNR ESPRIT Tradeoff between Accuracy and Computational intensity Frequency Independent Beamformer Beamforming is the signal processing technique which operate on multiple sensor arrays Types of Beamforming Frequency Dependent Frequency Independent Conventional Beamformers are all frequency dependent. The few Frequency independent beamformers available work with large(512 microphone) array systems Novel approach The algorithm developed at CNEL works on a narrow baseline (4cm) 2 microphone system The results are superior to conventional techniques Speech with babble noise in the background Signal processed by the algorithm Speech with pink noise in the background Signal processed by the algorithm Improvements in SNRImprovement in Recognition Results Improvements in SNR for varying Noise DOA Performance comparison between Motorola's Noise suppressor and our algorithm Outperforms!