Automatic Speech Codec Identification with Applications To
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Omtp Codecs 1 0, Release 1
OMTP CODECS DEFINITION AND REQUIREMENTS This document contains information that is confidential and proprietary to OMTP Limited. The information may not be used, disclosed or reproduced without the prior written authorisation of OMTP Limited, and those so authorised may only use this information for the purpose consistent with the authorisation. VERSION: OMTP CODECS 1_0, RELEASE 1 STATUS: SUBJECT TO BE - APPROVED BY BOARD 21 JULY 2005 DATE OF LAST EDIT: 6 JULY 2005 OWNER: P4: HARDWARE REQUIREMENTS AND DE- FRAGMENTATION OMTP CODECS CONTENTS 1 INTRODUCTION ............................................................................4 1.1 DOCUMENT PURPOSE ..........................................................................4 1.2 INTENDED AUDIENCE ............................................................................5 2 DEFINITION OF TERMS .................................................................7 2.1 CONVENTIONS .....................................................................................7 3 OMTP CODEC PROFILES ............................................................8 3.1 AUDIO DECODE....................................................................................8 3.2 AUDIO ENCODE....................................................................................9 3.3 VIDEO DECODE....................................................................................9 3.4 VIDEO ENCODE....................................................................................9 3.5 IMAGE DECODE..................................................................................10 -
Packetcable™ 2.0 Codec and Media Specification PKT-SP-CODEC
PacketCable™ 2.0 Codec and Media Specification PKT-SP-CODEC-MEDIA-I10-120412 ISSUED Notice This PacketCable specification is the result of a cooperative effort undertaken at the direction of Cable Television Laboratories, Inc. for the benefit of the cable industry and its customers. This document may contain references to other documents not owned or controlled by CableLabs. Use and understanding of this document may require access to such other documents. Designing, manufacturing, distributing, using, selling, or servicing products, or providing services, based on this document may require intellectual property licenses from third parties for technology referenced in this document. Neither CableLabs nor any member company is responsible to any party for any liability of any nature whatsoever resulting from or arising out of use or reliance upon this document, or any document referenced herein. This document is furnished on an "AS IS" basis and neither CableLabs nor its members provides any representation or warranty, express or implied, regarding the accuracy, completeness, noninfringement, or fitness for a particular purpose of this document, or any document referenced herein. 2006-2012 Cable Television Laboratories, Inc. All rights reserved. PKT-SP-CODEC-MEDIA-I10-120412 PacketCable™ 2.0 Document Status Sheet Document Control Number: PKT-SP-CODEC-MEDIA-I10-120412 Document Title: Codec and Media Specification Revision History: I01 - Released 04/05/06 I02 - Released 10/13/06 I03 - Released 09/25/07 I04 - Released 04/25/08 I05 - Released 07/10/08 I06 - Released 05/28/09 I07 - Released 07/02/09 I08 - Released 01/20/10 I09 - Released 05/27/10 I10 – Released 04/12/12 Date: April 12, 2012 Status: Work in Draft Issued Closed Progress Distribution Restrictions: Authors CL/Member CL/ Member/ Public Only Vendor Key to Document Status Codes: Work in Progress An incomplete document, designed to guide discussion and generate feedback, that may include several alternative requirements for consideration. -
Influence of Speech Codecs Selection on Transcoding Steganography
Influence of Speech Codecs Selection on Transcoding Steganography Artur Janicki, Wojciech Mazurczyk, Krzysztof Szczypiorski Warsaw University of Technology, Institute of Telecommunications Warsaw, Poland, 00-665, Nowowiejska 15/19 Abstract. The typical approach to steganography is to compress the covert data in order to limit its size, which is reasonable in the context of a limited steganographic bandwidth. TranSteg (Trancoding Steganography) is a new IP telephony steganographic method that was recently proposed that offers high steganographic bandwidth while retaining good voice quality. In TranSteg, compression of the overt data is used to make space for the steganogram. In this paper we focus on analyzing the influence of the selection of speech codecs on hidden transmission performance, that is, which codecs would be the most advantageous ones for TranSteg. Therefore, by considering the codecs which are currently most popular for IP telephony we aim to find out which codecs should be chosen for transcoding to minimize the negative influence on voice quality while maximizing the obtained steganographic bandwidth. Key words: IP telephony, network steganography, TranSteg, information hiding, speech coding 1. Introduction Steganography is an ancient art that encompasses various information hiding techniques, whose aim is to embed a secret message (steganogram) into a carrier of this message. Steganographic methods are aimed at hiding the very existence of the communication, and therefore any third-party observers should remain unaware of the presence of the steganographic exchange. Steganographic carriers have evolved throughout the ages and are related to the evolution of the methods of communication between people. Thus, it is not surprising that currently telecommunication networks are a natural target for steganography. -
CT8021 H.32X G.723.1/G.728 Truespeech Co-Processor
CT8021 H.32x G.723.1/G.728 TrueSpeech Co-Processor Introduction Features The CT8021 is a speech co-processor which · TrueSpeechâ G.723.1 at 6.3, 5.3, 4.8 and performs full duplex speech compression and de- 4.1 kbps at 8KHz sampling rate (including compression functions. It provides speech G.723.1 Annex A VAD/CNG) compression for H.320, H.323 and H.324 · G.728 16 Kbps LD-CELP Multimedia Visual Telephony / Video · Download of additional speech compression Conferencing products and DSVD Modems. The software modules into external sram for CT8021 has built-in TrueSpeechâ G.723.1 (for TrueSpeechâ 8.5, G.722 & G.729-A/B H.323 and H.324) as well as G.728 LD-CELP · Real-time Full duplex or Half duplex speech speech compression (for H.320). This combination compression and decompression of ITU speech compression standards within a · Acoustic Echo Cancellation concurrent with single device enables the creation of a single full-duplex speech compression multimedia terminal which can operate in all · Full Duplex standalone Speakerphone types of Video Conferencing systems including · Host-to-Host (codec-less) and Host-CODEC H.320 ISDN-based, H.324 POTS-based, and modes of operation H.323 LAN/Internet-based. TrueSpeechâ G.723.1 · Parallel 8-bit host interface provides simple provides compressed data rates of 6.3 and 5.3 memory-mapped I/O host connection. Kbps and includes G.723.1 Annex A VAD/CNG · 1 or 2-channel DMA support (Single Cycle “silence” compression which can supply an even and Burst Modes) lower average bit rate. -
TR 101 329-7 V1.1.1 (2000-11) Technical Report
ETSI TR 101 329-7 V1.1.1 (2000-11) Technical Report TIPHON; Design Guide; Part 7: Design Guide for Elements of a TIPHON connection from an end-to-end speech transmission performance point of view 2 ETSI TR 101 329-7 V1.1.1 (2000-11) Reference DTR/TIPHON-05011 Keywords internet, IP, network, performance, protocol, quality, speech, voice ETSI 650 Route des Lucioles F-06921 Sophia Antipolis Cedex - FRANCE Tel.:+33492944200 Fax:+33493654716 Siret N° 348 623 562 00017 - NAF 742 C Association à but non lucratif enregistrée à la Sous-Préfecture de Grasse (06) N° 7803/88 Important notice Individual copies of the present document can be downloaded from: http://www.etsi.org The present document may be made available in more than one electronic version or in print. In any case of existing or perceived difference in contents between such versions, the reference version is the Portable Document Format (PDF). In case of dispute, the reference shall be the printing on ETSI printers of the PDF version kept on a specific network drive within ETSI Secretariat. Users of the present document should be aware that the document may be subject to revision or change of status. Information on the current status of this and other ETSI documents is available at http://www.etsi.org/tb/status/ If you find errors in the present document, send your comment to: [email protected] Copyright Notification No part may be reproduced except as authorized by written permission. The copyright and the foregoing restriction extend to reproduction in all media. -
Multiple Description Coding Using Time Domain Division for MP3 Coded Sound Signal
Journal of Information Hiding and Multimedia Signal Processing ©2010 ISSN 2073-4212 Ubiquitous International Volume 1, Number 4, October 2010 Multiple Description Coding Using Time Domain Division for MP3 coded Sound Signal Ho-seok Wey1;2, Akinori Ito3, Takuma Okamoto1;3, and Yoiti Suzuki1;3 1Research Institute of Electrical Communication,Tohoku University 2-1-1, Katahira, Aoba-ku, Sendai, Miyagi 980-8577, JAPAN 2Graduate School of Information Sciences, Tohoku University, JAPAN Tel.: +81-22-217-5461, Fax.: +81-22-217-5535, E-mail: fhswey, okamoto, yoh @ais.riec.tohoku.ac.jp 3Graduate School of Engineering,f Tohoku University, g 6-6-05 Aramaki aza Aoba, Aoba-ku, Sendai, Miyagi 980-8579, JAPAN Tel.,Fax.: +81-22-795-7084, E-mail: [email protected] Received April 2010; revised May 2010 Abstract. In audio communications over a lossy packet network, packet loss conceal- ment techniques are needed to mitigate a user's frustration when perceiving the deterio- ration of the quality of the decoded signal. Multiple description coding (MDC) is a useful solution to this problem. In this paper, we describe an MDC method for concealing packet losses for wideband sound signal streaming based on the sample splitting method in the time domain and encoding by an MPEG-1 audio layer III (MP3) encoder. To enhance the quality of the restored signal, we applied a Wiener filter to the higher frequency part of the restored signal. Experiments were conducted to compare the proposed method with several conventional methods, conrming that the proposed method showed higher quality results than the conventional methods for a range of bit rates from 128 to 192 kbps when there were heavy packet losses. -
Essential Technology for High-Compression Speech Coding Takehiro Moriya
Information LSP (Line Spectrum Pair): Essential Technology for High-compression Speech Coding Takehiro Moriya Abstract Line Spectrum Pair (LSP) technology was accepted as an IEEE (Institute of Electrical and Electronics Engineers) Milestone in 204. LSP, invented by Dr. Fumitada Itakura at NTT in 975, is an efficient method for representing speech spectra, namely, the shape of the vocal tract. A speech synthesis large-scale integration chip based on LSP was fabricated in 980. Since the 990s, LSP has been adopted in many speech coding standards as an essential component, and it is still used worldwide in almost all cellular phones and Internet protocol phones. Keywords: LSP, speech coding, cellular phone 1. Introduction President and CEO of NTT (Photo 2), at a ceremony held in Tokyo. The citation reads, “Line Spectrum On May 22, 204, Line Spectrum Pair (LSP) tech- Pair (LSP) for high-compression speech coding, nology was officially recognized as an Institute of 975. Line Spectrum Pair, invented at NTT in 975, Electrical and Electronics Engineers (IEEE) Mile- is an important technology for speech synthesis and stone. Dr. J. Roberto de Marca, President of IEEE, coding. A speech synthesizer chip was designed presented the plaque (Photo 1) to Mr. Hiroo Unoura, based on Line Spectrum Pair in 980. In the 990s, Photo 1. Plaque of IEEE Milestone for Line Spectrum Pair (LSP) for high-compression speech coding. Photo 2. From IEEE president to NTT president. NTT Technical Review Information this technology was adopted in almost all interna- better quantization and interpolation performance tional speech coding standards as an essential compo- than PARCOR. -
A Real-Time Audio Compression Technique Based on Fast Wavelet Filtering and Encoding
Proceedings of the Federated Conference on Computer Science DOI: 10.15439/2016F296 and Information Systems pp. 497–502 ACSIS, Vol. 8. ISSN 2300-5963 A Real-Time Audio Compression Technique Based on Fast Wavelet Filtering and Encoding Nella Romano∗, Antony Scivoletto∗, Dawid Połap† ∗Department of Electrical and Informatics Engineering, University of Catania, Viale A. Doria 6, 95125 Catania, Italy Email: [email protected], [email protected] †Institute of Mathematics, Silesian University of Technology, Kaszubska 23, 44-100 Gliwice, Poland Email: [email protected] Abstract—With the development of telecommunication tech- transmitted using less data than would be required for the nology over the last decades, the request for digital information original signal [2]. compression has increased dramatically. In many applications, Compression techniques can be classified into one of two such as high quality audio transmission and storage, the target is to achieve audio and speech signal codings at the lowest main categories: lossless and lossy. Lossless compression possible data rates, in order to offer cheaper costs in terms works by removing the redundant information present in an of transmission and storage. Recently, compression techniques audio signal, but preserving its quality and the complete in- using wavelet transform have received great attention because tegrity of the data. However, it offers small compression ratios, of their promising compression ratio, signal to noise ratio, hence it can be used if we have no stringent requirements; and flexibility in representing speech signals. In this paper we examine a new technique for analysing and compressing furthermore, it does not guarantee a constant output data rate, speech signals using biorthogonal wavelet filters. -
Low Rate Speech Coding and Random Bit Errors: a Subjective Speech Quality Matching Experiment
NTIA Technical Report TR-10-462 Low Rate Speech Coding and Random Bit Errors: A Subjective Speech Quality Matching Experiment Andrew A. Catellier Stephen D. Voran NTIA Technical Report TR-10-462 Low Rate Speech Coding and Random Bit Errors: A Subjective Speech Quality Matching Experiment Andrew A. Catellier Stephen D. Voran U.S. DEPARTMENT OF COMMERCE Gary Locke, Secretary Lawrence E. Strickling, Assistant Secretary for Communications and Information October 2009 DISCLAIMER Certain commercial equipment, instruments, or materials are identified in this report to adequately describe the experimental procedure. In no case does such identification imply recommendation or endorsement by the National Telecommunications and Information Administration, nor does it imply that the material or equipment identified is necessarily the best available for the purpose. iii CONTENTS Page FIGURES ............................................ vi TABLES ............................................. vii 1 INTRODUCTION ..................................... 1 2 EXPERIMENT DESCRIPTION .............................. 3 2.1 Speech Recordings . 3 2.2 Experiment Procedure . 4 2.3 Listeners . 5 2.4 Speech Quality Matching Algorithm . 5 2.5 Speech Coding and Bit Errors . 7 2.6 Bookkeeping . 10 2.7 Objective Speech Quality Estimation . 10 3 RESULTS .......................................... 13 3.1 Listening Experiment . 13 3.2 Objective Estimation . 14 3.3 Bit Error Statistics . 16 4 CONCLUSIONS ...................................... 18 5 ACKNOWLEDGEMENTS ................................. 19 6 REFERENCES ....................................... 20 v FIGURES Page Figure 1. The human interface used during the experiment. 5 Figure 2. Signal flow through the encoding, bit error, and decoding processes. 7 Figure 3. Example bit error patterns that conform with the subset property. BER is 15, 10, and 5% in the top, middle and bottom panels, respectively. 9 Figure 4. -
Ts 148 060 V13.0.0 (2016-01)
ETSI TS 1148 060 V13.0.0 (201616-01) TECHNICAL SPECIFICATIONION Digital cellular telecocommunications system (Phahase 2+); In-band control of remremote transcoders and rate aadaptors for fullull rate traffic channels (3GPP TS 48.0.060 version 13.0.0 Release 13) R GLOBAL SYSTETEM FOR MOBILE COMMUNUNICATIONS 3GPP TS 48.060 version 13.0.0 Release 13 1 ETSI TS 148 060 V13.0.0 (2016-01) Reference RTS/TSGG-0148060vd00 Keywords GSM ETSI 650 Route des Lucioles F-06921 Sophia Antipolis Cedex - FRANCE Tel.: +33 4 92 94 42 00 Fax: +33 4 93 65 47 16 Siret N° 348 623 562 00017 - NAF 742 C Association à but non lucratif enregistrée à la Sous-Préfecture de Grasse (06) N° 7803/88 Important notice The present document can be downloaded from: http://www.etsi.org/standards-search The present document may be made available in electronic versions and/or in print. The content of any electronic and/or print versions of the present document shall not be modified without the prior written authorization of ETSI. In case of any existing or perceived difference in contents between such versions and/or in print, the only prevailing document is the print of the Portable Document Format (PDF) version kept on a specific network drive within ETSI Secretariat. Users of the present document should be aware that the document may be subject to revision or change of status. Information on the current status of this and other ETSI documents is available at http://portal.etsi.org/tb/status/status.asp If you find errors in the present document, please send your comment to one of the following services: https://portal.etsi.org/People/CommiteeSupportStaff.aspx Copyright Notification No part may be reproduced or utilized in any form or by any means, electronic or mechanical, including photocopying and microfilm except as authorized by written permission of ETSI. -
A Cross-Layer Design for Scalable Mobile Video
A Cross-Layer Design for Scalable Mobile Video The MIT Faculty has made this article openly available. Please share how this access benefits you. Your story matters. Citation Jakubczak, Szymon, and Dina Katabi. “A Cross-layer Design for Scalable Mobile Video.” ACM Press, 2011. 289. As Published http://dx.doi.org/10.1145/2030613.2030646 Publisher Association for Computing Machinery (ACM) Version Author's final manuscript Citable link http://hdl.handle.net/1721.1/72994 Terms of Use Creative Commons Attribution-Noncommercial-Share Alike 3.0 Detailed Terms http://creativecommons.org/licenses/by-nc-sa/3.0/ A Cross-Layer Design for Scalable Mobile Video Szymon Jakubczak Dina Katabi CSAIL MIT CSAIL MIT 32 Vassar St. 32 Vassar St. Cambridge, Mass. 02139 Cambridge, Mass. 02139 [email protected] [email protected] ABSTRACT Cisco visual index, mobile video traffic will grow 66 fold over Today’s mobile video suffers from two limitations: 1) it can- a period of five years [1]. Such predictions lead to a natu- not reduce bandwidth consumption by leveraging wireless ral question: can existing wireless technologies, e.g., WiFi, broadcast to multicast popular content to interested re- WiMax, or LTE, support this impending demand and pro- ceivers, and 2) it lacks robustness to wireless interference vide scalable and robust mobile video? and errors. This paper presents SoftCast, a cross-layer de- (a) Scalability. As demands for mobile video increase sign for mobile video that addresses both limitations. To congestion will also increase. The problem becomes particu- do so, SoftCast changes the network stack to act like a lin- larly severe when many users try to watch a popular realtime ear transform. -
AMR Wideband Speech Codec; Feasibility Study Report (3GPP TR 26.901 Version 4.0.1 Release 4)
ETSI TR 126 901 V4.0.1 (2001-04) Technical Report Universal Mobile Telecommunications System (UMTS); AMR wideband speech codec; Feasibility study report (3GPP TR 26.901 version 4.0.1 Release 4) 3GPP TR 26.901 version 4.0.1 Release 4 1 ETSI TR 126 901 V4.0.1 (2001-04) Reference DTR/TSGS-0426901Uv4 Keywords UMTS ETSI 650 Route des Lucioles F-06921 Sophia Antipolis Cedex - FRANCE Tel.:+33492944200 Fax:+33493654716 Siret N° 348 623 562 00017 - NAF 742 C Association à but non lucratif enregistrée à la Sous-Préfecture de Grasse (06) N° 7803/88 Important notice Individual copies of the present document can be downloaded from: http://www.etsi.org The present document may be made available in more than one electronic version or in print. In any case of existing or perceived difference in contents between such versions, the reference version is the Portable Document Format (PDF). In case of dispute, the reference shall be the printing on ETSI printers of the PDF version kept on a specific network drive within ETSI Secretariat. Users of the present document should be aware that the document may be subject to revision or change of status. Information on the current status of this and other ETSI documents is available at http://www.etsi.org/tb/status/ If you find errors in the present document, send your comment to: [email protected] Copyright Notification No part may be reproduced except as authorized by written permission. The copyright and the foregoing restriction extend to reproduction in all media. © European Telecommunications Standards Institute 2001.