Fast Computation of LSP Frequencies Using the Bairstow Method
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Digital Speech Processing— Lecture 17
Digital Speech Processing— Lecture 17 Speech Coding Methods Based on Speech Models 1 Waveform Coding versus Block Processing • Waveform coding – sample-by-sample matching of waveforms – coding quality measured using SNR • Source modeling (block processing) – block processing of signal => vector of outputs every block – overlapped blocks Block 1 Block 2 Block 3 2 Model-Based Speech Coding • we’ve carried waveform coding based on optimizing and maximizing SNR about as far as possible – achieved bit rate reductions on the order of 4:1 (i.e., from 128 Kbps PCM to 32 Kbps ADPCM) at the same time achieving toll quality SNR for telephone-bandwidth speech • to lower bit rate further without reducing speech quality, we need to exploit features of the speech production model, including: – source modeling – spectrum modeling – use of codebook methods for coding efficiency • we also need a new way of comparing performance of different waveform and model-based coding methods – an objective measure, like SNR, isn’t an appropriate measure for model- based coders since they operate on blocks of speech and don’t follow the waveform on a sample-by-sample basis – new subjective measures need to be used that measure user-perceived quality, intelligibility, and robustness to multiple factors 3 Topics Covered in this Lecture • Enhancements for ADPCM Coders – pitch prediction – noise shaping • Analysis-by-Synthesis Speech Coders – multipulse linear prediction coder (MPLPC) – code-excited linear prediction (CELP) • Open-Loop Speech Coders – two-state excitation -
Mpeg Vbr Slice Layer Model Using Linear Predictive Coding and Generalized Periodic Markov Chains
MPEG VBR SLICE LAYER MODEL USING LINEAR PREDICTIVE CODING AND GENERALIZED PERIODIC MARKOV CHAINS Michael R. Izquierdo* and Douglas S. Reeves** *Network Hardware Division IBM Corporation Research Triangle Park, NC 27709 [email protected] **Electrical and Computer Engineering North Carolina State University Raleigh, North Carolina 27695 [email protected] ABSTRACT The ATM Network has gained much attention as an effective means to transfer voice, video and data information We present an MPEG slice layer model for VBR over computer networks. ATM provides an excellent vehicle encoded video using Linear Predictive Coding (LPC) and for video transport since it provides low latency with mini- Generalized Periodic Markov Chains. Each slice position mal delay jitter when compared to traditional packet net- within an MPEG frame is modeled using an LPC autoregres- works [11]. As a consequence, there has been much research sive function. The selection of the particular LPC function is in the area of the transmission and multiplexing of com- governed by a Generalized Periodic Markov Chain; one pressed video data streams over ATM. chain is defined for each I, P, and B frame type. The model is Compressed video differs greatly from classical packet sufficiently modular in that sequences which exclude B data sources in that it is inherently quite bursty. This is due to frames can eliminate the corresponding Markov Chain. We both temporal and spatial content variations, bounded by a show that the model matches the pseudo-periodic autocorre- fixed picture display rate. Rate control techniques, such as lation function quite well. We present simulation results of CBR (Constant Bit Rate), were developed in order to reduce an Asynchronous Transfer Mode (ATM) video transmitter the burstiness of a video stream. -
Relating Optical Speech to Speech Acoustics and Visual Speech Perception
UNIVERSITY OF CALIFORNIA Los Angeles Relating Optical Speech to Speech Acoustics and Visual Speech Perception A dissertation submitted in partial satisfaction of the requirements for the degree Doctor of Philosophy in Electrical Engineering by Jintao Jiang 2003 i © Copyright by Jintao Jiang 2003 ii The dissertation of Jintao Jiang is approved. Kung Yao Lieven Vandenberghe Patricia A. Keating Lynne E. Bernstein Abeer Alwan, Committee Chair University of California, Los Angeles 2003 ii Table of Contents Chapter 1. Introduction ............................................................................................... 1 1.1. Audio-Visual Speech Processing ............................................................................ 1 1.2. How to Examine the Relationship between Data Sets ............................................ 4 1.3. The Relationship between Articulatory Movements and Speech Acoustics........... 5 1.4. The Relationship between Visual Speech Perception and Physical Measures ....... 9 1.5. Outline of This Dissertation .................................................................................. 15 Chapter 2. Data Collection and Pre-Processing ...................................................... 17 2.1. Introduction ........................................................................................................... 17 2.2. Background ........................................................................................................... 18 2.3. Recording a Database for the Correlation Analysis............................................. -
Improving Opus Low Bit Rate Quality with Neural Speech Synthesis
Improving Opus Low Bit Rate Quality with Neural Speech Synthesis Jan Skoglund1, Jean-Marc Valin2∗ 1Google, San Francisco, CA, USA 2Amazon, Palo Alto, CA, USA [email protected], [email protected] Abstract learned representation set [11]. A typical WaveNet configura- The voice mode of the Opus audio coder can compress wide- tion requires a very high algorithmic complexity, in the order band speech at bit rates ranging from 6 kb/s to 40 kb/s. How- of hundreds of GFLOPS, along with a high memory usage to ever, Opus is at its core a waveform matching coder, and as the hold the millions of model parameters. Combined with the high rate drops below 10 kb/s, quality degrades quickly. As the rate latency, in the hundreds of milliseconds, this renders WaveNet reduces even further, parametric coders tend to perform better impractical for a real-time implementation. Replacing the di- than waveform coders. In this paper we propose a backward- lated convolutional networks with recurrent networks improved compatible way of improving low bit rate Opus quality by re- memory efficiency in SampleRNN [12], which was shown to be synthesizing speech from the decoded parameters. We compare useful for speech coding in [13]. WaveRNN [14] also demon- two different neural generative models, WaveNet and LPCNet. strated possibilities for synthesizing at lower complexities com- WaveNet is a powerful, high-complexity, and high-latency ar- pared to WaveNet. Even lower complexity and real-time opera- chitecture that is not feasible for a practical system, yet pro- tion was recently reported using LPCNet [15]. vides a best known achievable quality with generative models. -
Lecture 16: Linear Prediction-Based Representations
Return to Main LECTURE 16: LINEAR PREDICTION-BASED REPRESENTATIONS Objectives ● Objectives: System View: Analysis/Synthesis ❍ Introduce analysis/synthesis systems The Error Signal ❍ Discuss error analysis and gain-matching Gain Matching ❍ Relate linear prediction coefficients to other spectral representations Transformations: Levinson-Durbin ❍ Introduce reflection and prediction coefficent recursions Reflection Coefficients Transformations ❍ Lattice/ladder filter implementations The Burg Method There is a classic textbook on this subject: Summary: Signal Modeling J.D. Markel and A.H. Gray, Linear Prediction of Speech, Springer-Verlag, New Typical Front End York, New York, USA, ISBN: 0-13-007444-6, 1976. On-Line Resources: This lecture also includes material from two other textbooks: AJR: Linear Prediction LPC10E MAD LPC J. Deller, et. al., Discrete-Time Processing of Speech Signals, MacMillan Publishing Co., ISBN: 0-7803-5386-2, 2000. Filter Design and, L.R. Rabiner and B.W. Juang, Fundamentals of Speech Recognition, Prentice-Hall, Upper Saddle River, New Jersey, USA, ISBN: 0-13-015157-2, 1993. Return to Main Introduction: 01: Organization (html, pdf) Speech Signals: ECE 8463: FUNDAMENTALS OF SPEECH 02: Production (html, pdf) RECOGNITION 03: Digital Models Professor Joseph Picone (html, pdf) Department of Electrical and Computer Engineering Mississippi State University 04: Perception (html, pdf) email: [email protected] phone/fax: 601-325-3149; office: 413 Simrall 05: Masking URL: http://www.isip.msstate.edu/resources/courses/ece_8463 (html, pdf) Modern speech understanding systems merge interdisciplinary technologies from Signal Processing, 06: Phonetics and Phonology Pattern Recognition, Natural Language, and Linguistics into a unified statistical framework. These (html, pdf) systems, which have applications in a wide range of signal processing problems, represent a revolution in Digital Signal Processing (DSP). -
Source-Channel Coding for CELP Speech Coders
2G'S c5 Source-Channel Coding for CELP Speech Coders J.A. Asenstorfer B.Ma.Sc. (Hons),B.Sc.(Hons),8.8',M'E' University of Adelaide Department of Electrical and Electronic Engineering Thesis submitted for the Degree of Doctor of Philosophy. Septemb er 26, L994 A*ond p.] l'ìq t Contents 1 1 Speech Coders for NoisY Channels 1.1 Introduction 1 2 I.2 Thesis Goals and Historical Perspective 2 Linear Predictive Coders Background I I 2.1 Introduction ' 10 2.2 Linear Predictive Models . 2.3 Autocorrelation of SPeech 15 17 2.4 Generic Code Excited Linear Prediction 3 Pitch Estimation Based on Chaos Theory 23 23 3.1 Introduction . 3.1.1 A Novel Approach to Pitch Estimation 25 3.2 The Non-Linear DYnamical Model 26 29 3.3 Determining the DelaY 32 3.4 Speech Database 3.5 Initial Findings 32 3.5.1 Poincaré Sections 34 35 3.5.2 Poincaré Section in Time 39 3.6 SpeechClassifrcation 40 3.6.1 The Zero Set 4l 3.6.2 The Devil's Staircase 42 3.6.3 Orbit Direction Change Indicator 45 3.7 Voicing Estimator 47 3.8 Pitch Tracking 49 3.9 The Pitch Estimation Algorithm ' 51 3.10 Results from the Pitch Estimator 55 3.11 Waveform Substitution 58 3.11.1 Replication of the Last Pitch Period 59 3.1I.2 Unvoiced Substitution 60 3.11.3 SPlicing Coders 61 3.11.4 Memory Considerations for CELP 61 3.12 Conclusion Data Problem 63 4 Vector Quantiser and the Missing 63 4.I Introduction bÐ 4.2 WhY Use Vector Quantisation? a Training Set 66 4.2.1 Generation of a Vector Quantiser Using Results 68 4.2.2 Some Useful Optimal Vector Quantisation ,7r 4.3 Heuristic Argument ùn . -
Samia Dawood Shakir
MODELLING TALKING HUMAN FACES Thesis submitted to Cardiff University in candidature for the degree of Doctor of Philosophy. Samia Dawood Shakir School of Engineering Cardiff University 2019 ii DECLARATION This work has not been submitted in substance for any other degree or award at this or any other university or place of learning, nor is being submitted concurrently in candidature for any degree or other award. Signed . (candidate) Date . STATEMENT 1 This thesis is being submitted in partial fulfillment of the requirements for the degree of PhD. Signed . (candidate) Date . STATEMENT 2 This thesis is the result of my own independent work/investigation, except where otherwise stated, and the thesis has not been edited by a third party beyond what is permitted by Cardiff Universitys Policy on the Use of Third Party Editors by Research Degree Students. Other sources are acknowledged by explicit references. The views expressed are my own. Signed . (candidate) Date . STATEMENT 3 I hereby give consent for my thesis, if accepted, to be available online in the Universitys Open Access repository and for inter-library loan, and for the title and summary to be made available to outside organisations. Signed . (candidate) Date . ABSTRACT This thesis investigates a number of new approaches for visual speech synthesis using data-driven methods to implement a talking face. The main contributions in this thesis are the following. The ac- curacy of shared Gaussian process latent variable model (SGPLVM) built using the active appearance model (AAM) and relative spectral transform-perceptual linear prediction (RASTAPLP) features is im- proved by employing a more accurate AAM. -
Speech Compression
information Review Speech Compression Jerry D. Gibson Department of Electrical and Computer Engineering, University of California, Santa Barbara, CA 93118, USA; [email protected]; Tel.: +1-805-893-6187 Academic Editor: Khalid Sayood Received: 22 April 2016; Accepted: 30 May 2016; Published: 3 June 2016 Abstract: Speech compression is a key technology underlying digital cellular communications, VoIP, voicemail, and voice response systems. We trace the evolution of speech coding based on the linear prediction model, highlight the key milestones in speech coding, and outline the structures of the most important speech coding standards. Current challenges, future research directions, fundamental limits on performance, and the critical open problem of speech coding for emergency first responders are all discussed. Keywords: speech coding; voice coding; speech coding standards; speech coding performance; linear prediction of speech 1. Introduction Speech coding is a critical technology for digital cellular communications, voice over Internet protocol (VoIP), voice response applications, and videoconferencing systems. In this paper, we present an abridged history of speech compression, a development of the dominant speech compression techniques, and a discussion of selected speech coding standards and their performance. We also discuss the future evolution of speech compression and speech compression research. We specifically develop the connection between rate distortion theory and speech compression, including rate distortion bounds for speech codecs. We use the terms speech compression, speech coding, and voice coding interchangeably in this paper. The voice signal contains not only what is said but also the vocal and aural characteristics of the speaker. As a consequence, it is usually desired to reproduce the voice signal, since we are interested in not only knowing what was said, but also in being able to identify the speaker. -
Advanced Speech Compression VIA Voice Excited Linear Predictive Coding Using Discrete Cosine Transform (DCT)
International Journal of Innovative Technology and Exploring Engineering (IJITEE) ISSN: 2278-3075, Volume-2 Issue-3, February 2013 Advanced Speech Compression VIA Voice Excited Linear Predictive Coding using Discrete Cosine Transform (DCT) Nikhil Sharma, Niharika Mehta Abstract: One of the most powerful speech analysis techniques LPC makes coding at low bit rates possible. For LPC-10, the is the method of linear predictive analysis. This method has bit rate is about 2.4 kbps. Even though this method results in become the predominant technique for representing speech for an artificial sounding speech, it is intelligible. This method low bit rate transmission or storage. The importance of this has found extensive use in military applications, where a method lies both in its ability to provide extremely accurate high quality speech is not as important as a low bit rate to estimates of the speech parameters and in its relative speed of computation. The basic idea behind linear predictive analysis is allow for heavy encryptions of secret data. However, since a that the speech sample can be approximated as a linear high quality sounding speech is required in the commercial combination of past samples. The linear predictor model provides market, engineers are faced with using other techniques that a robust, reliable and accurate method for estimating parameters normally use higher bit rates and result in higher quality that characterize the linear, time varying system. In this project, output. In LPC-10 vocal tract is represented as a time- we implement a voice excited LPC vocoder for low bit rate speech varying filter and speech is windowed about every 30ms. -
End-To-End Video-To-Speech Synthesis Using Generative Adversarial Networks
1 End-to-End Video-To-Speech Synthesis using Generative Adversarial Networks Rodrigo Mira, Konstantinos Vougioukas, Pingchuan Ma, Stavros Petridis Member, IEEE Bjorn¨ W. Schuller Fellow, IEEE, Maja Pantic Fellow, IEEE Video-to-speech is the process of reconstructing the audio speech from a video of a spoken utterance. Previous approaches to this task have relied on a two-step process where an intermediate representation is inferred from the video, and is then decoded into waveform audio using a vocoder or a waveform reconstruction algorithm. In this work, we propose a new end-to-end video-to-speech model based on Generative Adversarial Networks (GANs) which translates spoken video to waveform end-to-end without using any intermediate representation or separate waveform synthesis algorithm. Our model consists of an encoder-decoder architecture that receives raw video as input and generates speech, which is then fed to a waveform critic and a power critic. The use of an adversarial loss based on these two critics enables the direct synthesis of raw audio waveform and ensures its realism. In addition, the use of our three comparative losses helps establish direct correspondence between the generated audio and the input video. We show that this model is able to reconstruct speech with remarkable realism for constrained datasets such as GRID, and that it is the first end-to-end model to produce intelligible speech for LRW (Lip Reading in the Wild), featuring hundreds of speakers recorded entirely ‘in the wild’. We evaluate the generated samples in two different scenarios – seen and unseen speakers – using four objective metrics which measure the quality and intelligibility of artificial speech. -
Source Coding Basics and Speech Coding
Source Coding Basics and Speech Coding Yao Wang Polytechnic University, Brooklyn, NY11201 http://eeweb.poly.edu/~yao Outline • Why do we need to compress speech signals • Basic components in a source coding system • Variable length binary encoding: Huffman coding • Speech coding overview • Predictive coding: DPCM, ADPCM • Vocoder • Hybrid coding • Speech coding standards ©Yao Wang, 2006 EE3414: Speech Coding 2 Why do we need to compress? • Raw PCM speech (sampled at 8 kbps, represented with 8 bit/sample) has data rate of 64 kbps • Speech coding refers to a process that reduces the bit rate of a speech file • Speech coding enables a telephone company to carry more voice calls in a single fiber or cable • Speech coding is necessary for cellular phones, which has limited data rate for each user (<=16 kbps is desired!). • For the same mobile user, the lower the bit rate for a voice call, the more other services (data/image/video) can be accommodated. • Speech coding is also necessary for voice-over-IP, audio-visual teleconferencing, etc, to reduce the bandwidth consumption over the Internet ©Yao Wang, 2006 EE3414: Speech Coding 3 Basic Components in a Source Coding System Input Transformed Quantized Binary Samples parameters parameters bitstreams Transfor- Quanti- Binary mation zation Encoding Lossy Lossless Prediction Scalar Q Fixed length Transforms Vector Q Variable length Model fitting (Huffman, …... arithmetic, LZW) • Motivation for transformation --- To yield a more efficient representation of the original samples. ©Yao Wang, 2006 EE3414: -
Linear Predictive Modelling of Speech - Constraints and Line Spectrum Pair Decomposition
Helsinki University of Technology Laboratory of Acoustics and Audio Signal Processing Espoo 2004 Report 71 LINEAR PREDICTIVE MODELLING OF SPEECH - CONSTRAINTS AND LINE SPECTRUM PAIR DECOMPOSITION Tom Bäckström Dissertation for the degree of Doctor of Science in Technology to be presented with due permission for public examination and debate in Auditorium S4, Department of Electrical and Communications Engineering, Helsinki University of Technology, Espoo, Finland, on the 5th of March, 2004, at 12 o'clock noon. Helsinki University of Technology Department of Electrical and Communications Engineering Laboratory of Acoustics and Audio Signal Processing Teknillinen korkeakoulu Sähkö- ja tietoliikennetekniikan osasto Akustiikan ja äänenkäsittelytekniikan laboratorio Helsinki University of Technology Laboratory of Acoustics and Audio Signal Processing P.O. Box 3000 FIN-02015 HUT Tel. +358 9 4511 Fax +358 9 460 224 E-mail [email protected] ISBN 951-22-6946-5 ISSN 1456-6303 Otamedia Oy Espoo, Finland 2004 Let the floor be the limit! HELSINKI UNIVERSITY OF TECHNOLOGY ABSTRACT OF DOCTORAL DISSERTATION P.O. BOX 1000, FIN-02015 HUT http://www.hut.fi Author Tom Bäckström Name of the dissertation Linear predictive modelling for speech - Constraints and Line Spectrum Pair Decomposition Date of manuscript 6.2.2004 Date of the dissertation 5.3.2004 Monograph 4 Article dissertation (summary + original articles) Department Electrical and Communications Engineering Laboratory Laboratory Acoustics and Audio Signal Processing Field of research Speech processing Opponent(s) Associate Professor Peter Händel Supervisor Professor Paavo Alku (Instructor) Abstract In an exploration of the spectral modelling of speech, this thesis presents theory and applications of constrained linear predictive (LP) models.