Audio Coding
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Speech Coding and Compression
Introduction to voice coding/compression PCM coding Speech vocoders based on speech production LPC based vocoders Speech over packet networks Speech coding and compression Corso di Networked Multimedia Systems Master Universitario di Primo Livello in Progettazione e Gestione di Sistemi di Rete Carlo Drioli Università degli Studi di Verona Facoltà di Scienze Matematiche, Dipartimento di Informatica Fisiche e Naturali Speech coding and compression Introduction to voice coding/compression PCM coding Speech vocoders based on speech production LPC based vocoders Speech over packet networks Speech coding and compression: OUTLINE Introduction to voice coding/compression PCM coding Speech vocoders based on speech production LPC based vocoders Speech over packet networks Speech coding and compression Introduction to voice coding/compression PCM coding Speech vocoders based on speech production LPC based vocoders Speech over packet networks Introduction Approaches to voice coding/compression I Waveform coders (PCM) I Voice coders (vocoders) Quality assessment I Intelligibility I Naturalness (involves speaker identity preservation, emotion) I Subjective assessment: Listening test, Mean Opinion Score (MOS), Diagnostic acceptability measure (DAM), Diagnostic Rhyme Test (DRT) I Objective assessment: Signal to Noise Ratio (SNR), spectral distance measures, acoustic cues comparison Speech coding and compression Introduction to voice coding/compression PCM coding Speech vocoders based on speech production LPC based vocoders Speech over packet networks -
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 -
Audio Coding for Digital Broadcasting
Recommendation ITU-R BS.1196-7 (01/2019) Audio coding for digital broadcasting BS Series Broadcasting service (sound) ii Rec. ITU-R BS.1196-7 Foreword The role of the Radiocommunication Sector is to ensure the rational, equitable, efficient and economical use of the radio- frequency spectrum by all radiocommunication services, including satellite services, and carry out studies without limit of frequency range on the basis of which Recommendations are adopted. The regulatory and policy functions of the Radiocommunication Sector are performed by World and Regional Radiocommunication Conferences and Radiocommunication Assemblies supported by Study Groups. Policy on Intellectual Property Right (IPR) ITU-R policy on IPR is described in the Common Patent Policy for ITU-T/ITU-R/ISO/IEC referenced in Resolution ITU-R 1. Forms to be used for the submission of patent statements and licensing declarations by patent holders are available from http://www.itu.int/ITU-R/go/patents/en where the Guidelines for Implementation of the Common Patent Policy for ITU-T/ITU-R/ISO/IEC and the ITU-R patent information database can also be found. Series of ITU-R Recommendations (Also available online at http://www.itu.int/publ/R-REC/en) Series Title BO Satellite delivery BR Recording for production, archival and play-out; film for television BS Broadcasting service (sound) BT Broadcasting service (television) F Fixed service M Mobile, radiodetermination, amateur and related satellite services P Radiowave propagation RA Radio astronomy RS Remote sensing systems S Fixed-satellite service SA Space applications and meteorology SF Frequency sharing and coordination between fixed-satellite and fixed service systems SM Spectrum management SNG Satellite news gathering TF Time signals and frequency standards emissions V Vocabulary and related subjects Note: This ITU-R Recommendation was approved in English under the procedure detailed in Resolution ITU-R 1. -
Speech Compression Using Discrete Wavelet Transform and Discrete Cosine Transform
International Journal of Engineering Research & Technology (IJERT) ISSN: 2278-0181 Vol. 1 Issue 5, July - 2012 Speech Compression Using Discrete Wavelet Transform and Discrete Cosine Transform Smita Vatsa, Dr. O. P. Sahu M. Tech (ECE Student ) Professor Department of Electronicsand Department of Electronics and Communication Engineering Communication Engineering NIT Kurukshetra, India NIT Kurukshetra, India Abstract reproduced with desired level of quality. Main approaches of speech compression used today are Aim of this paper is to explain and implement waveform coding, transform coding and parametric transform based speech compression techniques. coding. Waveform coding attempts to reproduce Transform coding is based on compressing signal input signal waveform at the output. In transform by removing redundancies present in it. Speech coding at the beginning of procedure signal is compression (coding) is a technique to transform transformed into frequency domain, afterwards speech signal into compact format such that speech only dominant spectral features of signal are signal can be transmitted and stored with reduced maintained. In parametric coding signals are bandwidth and storage space respectively represented through a small set of parameters that .Objective of speech compression is to enhance can describe it accurately. Parametric coders transmission and storage capacity. In this paper attempts to produce a signal that sounds like Discrete wavelet transform and Discrete cosine original speechwhether or not time waveform transform -
Preview - Click Here to Buy the Full Publication
This is a preview - click here to buy the full publication IEC 62481-2 ® Edition 2.0 2013-09 INTERNATIONAL STANDARD colour inside Digital living network alliance (DLNA) home networked device interoperability guidelines – Part 2: DLNA media formats INTERNATIONAL ELECTROTECHNICAL COMMISSION PRICE CODE XH ICS 35.100.05; 35.110; 33.160 ISBN 978-2-8322-0937-0 Warning! Make sure that you obtained this publication from an authorized distributor. ® Registered trademark of the International Electrotechnical Commission This is a preview - click here to buy the full publication – 2 – 62481-2 © IEC:2013(E) CONTENTS FOREWORD ......................................................................................................................... 20 INTRODUCTION ................................................................................................................... 22 1 Scope ............................................................................................................................. 23 2 Normative references ..................................................................................................... 23 3 Terms, definitions and abbreviated terms ....................................................................... 30 3.1 Terms and definitions ............................................................................................ 30 3.2 Abbreviated terms ................................................................................................. 34 3.4 Conventions ......................................................................................................... -
Speech Coding Using Code Excited Linear Prediction
ISSN : 0976-8491 (Online) | ISSN : 2229-4333 (Print) IJCST VOL . 5, Iss UE SPL - 2, JAN - MAR C H 2014 Speech Coding Using Code Excited Linear Prediction 1Hemanta Kumar Palo, 2Kailash Rout 1ITER, Siksha ‘O’ Anusandhan University, Bhubaneswar, Odisha, India 2Gandhi Institute For Technology, Bhubaneswar, Odisha, India Abstract The main problem with the speech coding system is the optimum utilization of channel bandwidth. Due to this the speech signal is coded by using as few bits as possible to get low bit-rate speech coders. As the bit rate of the coder goes low, the intelligibility, SNR and overall quality of the speech signal decreases. Hence a comparative analysis is done of two different types of speech coders in this paper for understanding the utility of these coders in various applications so as to reduce the bandwidth and by different speech coding techniques and by reducing the number of bits without any appreciable compromise on the quality of speech. Hindi language has different number of stops than English , hence the performance of the coders must be checked on different languages. The main objective of this paper is to develop speech coders capable of producing high quality speech at low data rates. The focus of this paper is the development and testing of voice coding systems which cater for the above needs. Keywords PCM, DPCM, ADPCM, LPC, CELP Fig. 1: Speech Production Process I. Introduction Speech coding or speech compression is the compact digital The two types of speech sounds are voiced and unvoiced [1]. representations of voice signals [1-3] for the purpose of efficient They produce different sounds and spectra due to their differences storage and transmission. -
An Ultra Low-Power Miniature Speech CODEC at 8 Kb/S and 16 Kb/S Robert Brennan, David Coode, Dustin Griesdorf, Todd Schneider Dspfactory Ltd
To appear in ICSPAT 2000 Proceedings, October 16-19, 2000, Dallas, TX. An Ultra Low-power Miniature Speech CODEC at 8 kb/s and 16 kb/s Robert Brennan, David Coode, Dustin Griesdorf, Todd Schneider Dspfactory Ltd. 611 Kumpf Drive, Unit 200 Waterloo, Ontario, N2V 1K8 Canada Abstract SmartCODEC platform consists of an effi- cient, block-floating point, oversampled This paper describes a CODEC implementa- Weighted OverLap-Add (WOLA) filterbank, a tion on an ultra low-power miniature applica- software-programmable dual-Harvard 16-bit tion specific signal processor (ASSP) designed DSP core, two high fidelity 14-bit A/D con- for mobile audio signal processing applica- verters, a 14-bit D/A converter and a flexible tions. The CODEC records speech to and set of peripherals [1]. The system hardware plays back from a serial flash memory at data architecture (Figure 1) was designed to enable rates of 16 and 8 kb/s, with a bandwidth of 4 memory upgrades. Removable memory cards kHz. This CODEC consumes only 1 mW in a or more power-efficient memory could be package small enough for use in a range of substituted for the serial flash memory. demanding portable applications. Results, improvements and applications are also dis- The CODEC communicates with the flash cussed. memory over an integrated SPI port that can transfer data at rates up to 80 kb/s. For this application, the port is configured to block 1. Introduction transfer frame packets every 14 ms. e Speech coding is ubiquitous. Increasing de- c i WOLA Filterbank v mands for portability and fidelity coupled with A/D e and D the desire for reduced storage and bandwidth o i l Programmable d o r u utilization have increased the demand for and t D/A DSP Core A deployment of speech CODECs. -
Lossless Compression of Audio Data
CHAPTER 12 Lossless Compression of Audio Data ROBERT C. MAHER OVERVIEW Lossless data compression of digital audio signals is useful when it is necessary to minimize the storage space or transmission bandwidth of audio data while still maintaining archival quality. Available techniques for lossless audio compression, or lossless audio packing, generally employ an adaptive waveform predictor with a variable-rate entropy coding of the residual, such as Huffman or Golomb-Rice coding. The amount of data compression can vary considerably from one audio waveform to another, but ratios of less than 3 are typical. Several freeware, shareware, and proprietary commercial lossless audio packing programs are available. 12.1 INTRODUCTION The Internet is increasingly being used as a means to deliver audio content to end-users for en tertainment, education, and commerce. It is clearly advantageous to minimize the time required to download an audio data file and the storage capacity required to hold it. Moreover, the expec tations of end-users with regard to signal quality, number of audio channels, meta-data such as song lyrics, and similar additional features provide incentives to compress the audio data. 12.1.1 Background In the past decade there have been significant breakthroughs in audio data compression using lossy perceptual coding [1]. These techniques lower the bit rate required to represent the signal by establishing perceptual error criteria, meaning that a model of human hearing perception is Copyright 2003. Elsevier Science (USA). 255 AU rights reserved. 256 PART III / APPLICATIONS used to guide the elimination of excess bits that can be either reconstructed (redundancy in the signal) orignored (inaudible components in the signal). -
4. MPEG Layer-3 Audio Encoding
MPEG Layer-3 An introduction to MPEG Layer-3 K. Brandenburg and H. Popp Fraunhofer Institut für Integrierte Schaltungen (IIS) MPEG Layer-3, otherwise known as MP3, has generated a phenomenal interest among Internet users, or at least among those who want to download highly-compressed digital audio files at near-CD quality. This article provides an introduction to the work of the MPEG group which was, and still is, responsible for bringing this open (i.e. non-proprietary) compression standard to the forefront of Internet audio downloads. 1. Introduction The audio coding scheme MPEG Layer-3 will soon celebrate its 10th birthday, having been standardized in 1991. In its first years, the scheme was mainly used within DSP- based codecs for studio applications, allowing professionals to use ISDN phone lines as cost-effective music links with high sound quality. In 1995, MPEG Layer-3 was selected as the audio format for the digital satellite broadcasting system developed by World- Space. This was its first step into the mass market. Its second step soon followed, due to the use of the Internet for the electronic distribution of music. Here, the proliferation of audio material – coded with MPEG Layer-3 (aka MP3) – has shown an exponential growth since 1995. By early 1999, “.mp3” had become the most popular search term on the Web (according to http://www.searchterms.com). In 1998, the “MPMAN” (by Saehan Information Systems, South Korea) was the first portable MP3 player, pioneering the road for numerous other manufacturers of consumer electronics. “MP3” has been featured in many articles, mostly on the business pages of newspapers and periodicals, due to its enormous impact on the recording industry. -
The H.264 Advanced Video Coding (AVC) Standard
Whitepaper: The H.264 Advanced Video Coding (AVC) Standard What It Means to Web Camera Performance Introduction A new generation of webcams is hitting the market that makes video conferencing a more lifelike experience for users, thanks to adoption of the breakthrough H.264 standard. This white paper explains some of the key benefits of H.264 encoding and why cameras with this technology should be on the shopping list of every business. The Need for Compression Today, Internet connection rates average in the range of a few megabits per second. While VGA video requires 147 megabits per second (Mbps) of data, full high definition (HD) 1080p video requires almost one gigabit per second of data, as illustrated in Table 1. Table 1. Display Resolution Format Comparison Format Horizontal Pixels Vertical Lines Pixels Megabits per second (Mbps) QVGA 320 240 76,800 37 VGA 640 480 307,200 147 720p 1280 720 921,600 442 1080p 1920 1080 2,073,600 995 Video Compression Techniques Digital video streams, especially at high definition (HD) resolution, represent huge amounts of data. In order to achieve real-time HD resolution over typical Internet connection bandwidths, video compression is required. The amount of compression required to transmit 1080p video over a three megabits per second link is 332:1! Video compression techniques use mathematical algorithms to reduce the amount of data needed to transmit or store video. Lossless Compression Lossless compression changes how data is stored without resulting in any loss of information. Zip files are losslessly compressed so that when they are unzipped, the original files are recovered. -
Cognitive Speech Coding Milos Cernak, Senior Member, IEEE, Afsaneh Asaei, Senior Member, IEEE, Alexandre Hyafil
1 Cognitive Speech Coding Milos Cernak, Senior Member, IEEE, Afsaneh Asaei, Senior Member, IEEE, Alexandre Hyafil Abstract—Speech coding is a field where compression ear and undergoes a highly complex transformation paradigms have not changed in the last 30 years. The before it is encoded efficiently by spikes at the auditory speech signals are most commonly encoded with com- nerve. This great efficiency in information representation pression methods that have roots in Linear Predictive has inspired speech engineers to incorporate aspects of theory dating back to the early 1940s. This paper tries to cognitive processing in when developing efficient speech bridge this influential theory with recent cognitive studies applicable in speech communication engineering. technologies. This tutorial article reviews the mechanisms of speech Speech coding is a field where research has slowed perception that lead to perceptual speech coding. Then considerably in recent years. This has occurred not it focuses on human speech communication and machine because it has achieved the ultimate in minimizing bit learning, and application of cognitive speech processing in rate for transparent speech quality, but because recent speech compression that presents a paradigm shift from improvements have been small and commercial applica- perceptual (auditory) speech processing towards cognitive tions (e.g., cell phones) have been mostly satisfactory for (auditory plus cortical) speech processing. The objective the general public, and the growth of available bandwidth of this tutorial is to provide an overview of the impact has reduced requirements to compress speech even fur- of cognitive speech processing on speech compression and discuss challenges faced in this interdisciplinary speech ther. -
A Novel Speech Enhancement Approach Based on Modified Dct and Improved Pitch Synchronous Analysis
American Journal of Applied Sciences 11 (1): 24-37, 2014 ISSN: 1546-9239 ©2014 Science Publication doi:10.3844/ajassp.2014.24.37 Published Online 11 (1) 2014 (http://www.thescipub.com/ajas.toc) A NOVEL SPEECH ENHANCEMENT APPROACH BASED ON MODIFIED DCT AND IMPROVED PITCH SYNCHRONOUS ANALYSIS 1Balaji, V.R. and 2S. Subramanian 1Department of ECE, Sri Krishna College of Engineering and Technology, Coimbatore, India 2Department of CSE, Coimbatore Institute of Engineering and Technology, Coimbatore, India Received 2013-06-03, Revised 2013-07-17; Accepted 2013-11-21 ABSTRACT Speech enhancement has become an essential issue within the field of speech and signal processing, because of the necessity to enhance the performance of voice communication systems in noisy environment. There has been a number of research works being carried out in speech processing but still there is always room for improvement. The main aim is to enhance the apparent quality of the speech and to improve the intelligibility. Signal representation and enhancement in cosine transformation is observed to provide significant results. Discrete Cosine Transformation has been widely used for speech enhancement. In this research work, instead of DCT, Advanced DCT (ADCT) which simultaneous offers energy compaction along with critical sampling and flexible window switching. In order to deal with the issue of frame to frame deviations of the Cosine Transformations, ADCT is integrated with Pitch Synchronous Analysis (PSA). Moreover, in order to improve the noise minimization performance of the system, Improved Iterative Wiener Filtering approach called Constrained Iterative Wiener Filtering (CIWF) is used in this approach. Thus, a novel ADCT based speech enhancement using improved iterative filtering algorithm integrated with PSA is used in this approach.