Adaptive Multirate Wide Band Version 5.0 12 June 2020
Total Page:16
File Type:pdf, Size:1020Kb
Load more
Recommended publications
-
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. -
Space Application Development Board
Space Application Development Board The Space Application Development Board (SADB-C6727B) enables developers to implement systems using standard and custom algorithms on processor hardware similar to what would be deployed for space in a fully radiation hardened (rad-hard) design. The SADB-C6727B is devised to be similar to a space qualified rad-hardened system design using the TI SMV320C6727B-SP DSP and high-rel memories (SRAM, SDRAM, NOR FLASH) from Cobham. The VOCAL SADB- C6727B development board does not use these high reliability space qualified components but rather implements various common bus widths and memory types/speeds for algorithm development and performance evaluation. VOCAL Technologies, Ltd. www.vocal.com 520 Lee Entrance, Suite 202 Tel: (716) 688-4675 Buffalo, New York 14228 Fax: (716) 639-0713 Audio inputs are handled by a TI ADS1278 high speed multichannel analog-to-digital converter (ADC) which has a variant qualified for space operation. Audio outputs may be generated from digital signal Pulse Density Modulation (PDM) or Pulse Width Modulation (PWM) signals to avoid the need for a space qualified digital-to-analog converter (DAC) hardware or a traditional audio codec circuit. A commercial grade AIC3106 audio codec is provided for algorithm development and for comparison of audio performance to PDM/PWM generated audio signals. Digital microphones (PDM or I2S formats) can be sampled by the processor directly or by the AIC3106 audio codec (PDM format only). Digital audio inputs/outputs can also be connected to other processors (using the McASP signals directly). Both SRAM and SDRAM memories are supported via configuration resistors to allow for developing and benchmarking algorithms using either 16-bit wide or 32-bit wide busses (also via configuration resistors). -
Concatenation of Compression Codecs – the Need for Objective Evaluations
Concatenation of compression codecs – The need for objective evaluations C.J. Dalton (UKIB) In this article the Author considers, firstly, a hypothetical broadcast network in which compression equipments have replaced several existing functions – resulting in multiple-cascading. Secondly, he describes a similar network that has been optimized for compression technology. Picture-quality assessment methods – both conventional and new, fied for still-picture applications in the printing in- subjective and objective – are dustry and was adapted to motion video applica- discussed with the aim of providing tions, by encoding on a picture-by-picture basis. background information. Some Various proprietary versions of JPEG, with higher proposals are put forward for compression ratios, were introduced to reduce the objective evaluation together with storage requirements, and alternative systems based on Wavelets and Fractals have been imple- initial observations when concaten- mented. Due to the availability of integrated hard- ating (cascading) codecs of similar ware, motion JPEG is now widely used for off- and different types. and on-line editing, disc servers, slow-motion sys- tems, etc., but there is little standardization and few interfaces at the compressed level. The MPEG-1 compression standard was aimed at 1. Introduction progressively-scanned moving images and has been widely adopted for CD-ROM and similar ap- Bit-rate reduction (BRR) techniques – commonly plications; it has also found limited application for referred to as compression – are now firmly estab- broadcast video. The MPEG-2 system is specifi- lished in the latest generation of broadcast televi- cally targeted at the complete gamut of video sys- sion equipments. tems; for standard-definition television, the MP@ML codec – with compression ratios of 20:1 Original language: English Manuscript received 15/1/97. -
MIGRATING RADIO CALL-IN TALK SHOWS to WIDEBAND AUDIO Radio Is the Original Social Network
Established 1961 2013 WBA Broadcasters Clinic Madison, WI MIGRATING RADIO CALL-IN TALK SHOWS TO WIDEBAND AUDIO Radio is the original Social Network • Serves local or national audience • Allows real-time commentary from the masses • The telephone becomes the medium • Telephone technical factors have limited the appeal of the radio “Social Network” Telephones have changed over the years But Telephone Sound has not changed (and has gotten worse) This is very bad for Radio Why do phones sound bad? • System designed for efficiency not comfort • Sampling rate of 8kHz chosen for all calls • 4 kHz max response • Enough for intelligibility • Loses depth, nuance, personality • Listener fatigue Why do phones sound so bad ? (cont) • Low end of telephone calls have intentional high- pass filtering • Meant to avoid AC power hum pickup in phone lines • Lose 2-1/2 Octaves of speech audio on low end • Not relevant for digital Why Phones Sound bad (cont) Los Angeles Times -- January 10, 2009 Verizon Communications Inc., the second-biggest U.S. telephone company, plans to do away with traditional phone lines within seven years as it moves to carry all calls over the Internet. An Internet-based service can be maintained at a fraction of the cost of a phone network and helps Verizon offer a greater range of services, Stratton said. "We've built our business over the years with circuit-switched voice being our bread and butter...but increasingly, we are in the business of selling, basically, data connectivity," Chief Marketing Officer John Stratton said. VoIP -
Ffmpeg Documentation Table of Contents
ffmpeg Documentation Table of Contents 1 Synopsis 2 Description 3 Detailed description 3.1 Filtering 3.1.1 Simple filtergraphs 3.1.2 Complex filtergraphs 3.2 Stream copy 4 Stream selection 5 Options 5.1 Stream specifiers 5.2 Generic options 5.3 AVOptions 5.4 Main options 5.5 Video Options 5.6 Advanced Video options 5.7 Audio Options 5.8 Advanced Audio options 5.9 Subtitle options 5.10 Advanced Subtitle options 5.11 Advanced options 5.12 Preset files 6 Tips 7 Examples 7.1 Preset files 7.2 Video and Audio grabbing 7.3 X11 grabbing 7.4 Video and Audio file format conversion 8 Syntax 8.1 Quoting and escaping 8.1.1 Examples 8.2 Date 8.3 Time duration 8.3.1 Examples 8.4 Video size 8.5 Video rate 8.6 Ratio 8.7 Color 8.8 Channel Layout 9 Expression Evaluation 10 OpenCL Options 11 Codec Options 12 Decoders 13 Video Decoders 13.1 rawvideo 13.1.1 Options 14 Audio Decoders 14.1 ac3 14.1.1 AC-3 Decoder Options 14.2 ffwavesynth 14.3 libcelt 14.4 libgsm 14.5 libilbc 14.5.1 Options 14.6 libopencore-amrnb 14.7 libopencore-amrwb 14.8 libopus 15 Subtitles Decoders 15.1 dvdsub 15.1.1 Options 15.2 libzvbi-teletext 15.2.1 Options 16 Encoders 17 Audio Encoders 17.1 aac 17.1.1 Options 17.2 ac3 and ac3_fixed 17.2.1 AC-3 Metadata 17.2.1.1 Metadata Control Options 17.2.1.2 Downmix Levels 17.2.1.3 Audio Production Information 17.2.1.4 Other Metadata Options 17.2.2 Extended Bitstream Information 17.2.2.1 Extended Bitstream Information - Part 1 17.2.2.2 Extended Bitstream Information - Part 2 17.2.3 Other AC-3 Encoding Options 17.2.4 Floating-Point-Only AC-3 Encoding -
Surround Sound Processed by Opus Codec: a Perceptual Quality Assessment
28. Konferenz Elektronische Sprachsignalverarbeitung 2017, Saarbrücken SURROUND SOUND PROCESSED BY OPUS CODEC: APERCEPTUAL QUALITY ASSESSMENT Franziska Trojahn, Martin Meszaros, Michael Maruschke and Oliver Jokisch Hochschule für Telekommunikation Leipzig, Germany [email protected] Abstract: The article describes the first perceptual quality study of 5.1 surround sound that has been processed by the Opus codec standardised by the Internet Engineering Task Force (IETF). All listening sessions with up to five subjects took place in a slightly sound absorbing laboratory – simulating living room conditions. For the assessment we conducted a Degradation Category Rating (DCR) listening- opinion test according to ITU-T P.800 recommendation with stimuli for six channels at total bitrates between 96 kbit/s and 192 kbit/s as well as hidden references. A group of 27 naive listeners compared a total of 20 sound samples. The differences between uncompressed and degraded sound samples were rated on a five-point degradation category scale resulting in Degradation Mean Opinion Score (DMOS). The overall results show that the average quality correlates with the bitrates. The quality diverges for the individual test stimuli depending on the music characteristics. Under most circumstances, a bitrate of 128 kbit/s is sufficient to achieve acceptable quality. 1 Introduction Nowadays, a high number of different speech and audio codecs are implemented in several kinds of multimedia applications; including audio / video entertainment, broadcasting and gaming. In recent years the demand for low delay and high quality audio applications, such as remote real-time jamming and cloud gaming, has been increasing. Therefore, current research objectives do not only include close to natural speech or audio quality, but also the requirements of low bitrates and a minimum latency. -
Polycom Voice
Level 1 Technical – Polycom Voice Level 1 Technical Polycom Voice Contents 1 - Glossary .......................................................................................................................... 2 2 - Polycom Voice Networks ................................................................................................. 3 Polycom UC Software ...................................................................................................... 3 Provisioning ..................................................................................................................... 3 3 - Key Features (Desktop and Conference Phones) ............................................................ 5 OpenSIP Integration ........................................................................................................ 5 Microsoft Integration ........................................................................................................ 5 Lync Qualification ............................................................................................................ 5 Better Together over Ethernet ......................................................................................... 5 BroadSoft UC-One Integration ......................................................................................... 5 Conference Link / Conference Link2 ................................................................................ 6 Polycom Desktop Connector .......................................................................................... -
NTT DOCOMO Technical Journal
3GPP EVS Codec for Unrivaled Speech Quality and Future Audio Communication over VoLTE EVS Speech Coding Standardization NTT DOCOMO has been engaged in the standardization of Research Laboratories Kimitaka Tsutsumi the 3GPP EVS codec, which is designed specifically for Kei Kikuiri VoLTE to further enhance speech quality, and has contributed to establishing a far-sighted strategy for making the EVS codec cover a variety of future communication services. Journal NTT DOCOMO has also proposed technical solutions that provide speech quality as high as FM radio broadcasts and that achieve both high coding efficiency and high audio quality not possible with any of the state-of-the-art speech codecs. The EVS codec will drive the emergence of a new style of speech communication entertainment that will combine BGM, sound effects, and voice in novel ways for mobile users. 2 Technical Band (AMR-WB)* [2] that is used in also encode music at high levels of quality 1. Introduction NTT DOCOMO’s VoLTE and that sup- and efficiency for non-real-time services, The launch of Voice over LTE (VoL- port wideband speech with a sampling 3GPP experts agreed to adopt high re- TE) services and flat-rate voice service frequency*3 of 16 kHz. In contrast, EVS quirements in the EVS codec for music has demonstrated the importance of high- has been designed to support super-wide- despite its main target of real-time com- quality telephony service to mobile users. band*4 speech with a sampling frequen- munication. Furthermore, considering In line with this trend, the 3rd Genera- cy of 32 kHz thereby achieving speech that telephony services using AMR-WB tion Partnership Project (3GPP) complet- of FM-radio quality*5. -
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.