A High-Quality Speech and Audio Codec with Less Than 10 Ms Delay
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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. -
TR-135 Data Model for a TR-069 Enabled STB
TECHNICAL REPORT TR-135 Data Model for a TR-069 Enabled STB Version: Issue 1 Version Date: December 2007 © The Broadband Forum. All rights reserved. Data Model for a TR-069 Enabled STB TR-135 Notice The Broadband Forum is a non-profit corporation organized to create guidelines for broadband network system development and deployment. This Technical Report has been approved by members of the Forum. This document is not binding on the Broadband Forum, any of its members, or any developer or service provider. This document is subject to change, but only with approval of members of the Forum. This document is provided "as is," with all faults. Any person holding a copyright in this document, or any portion thereof, disclaims to the fullest extent permitted by law any representation or warranty, express or implied, including, but not limited to, (a) any warranty of merchantability, fitness for a particular purpose, non-infringement, or title; (b) any warranty that the contents of the document are suitable for any purpose, even if that purpose is known to the copyright holder; (c) any warranty that the implementation of the contents of the documentation will not infringe any third party patents, copyrights, trademarks or other rights. This publication may incorporate intellectual property. The Broadband Forum encourages but does not require declaration of such intellectual property. For a list of declarations made by Broadband Forum member companies, please see www.broadband-forum.org. December 2007 © The Broadband Forum. All rights reserved. -
The Perceptual Impact of Different Quantization Schemes in G.719
The perceptual impact of different quantization schemes in G.719 BOTE LIU Master’s Degree Project Stockholm, Sweden May 2013 XR-EE-SIP 2013:001 Abstract In this thesis, three kinds of quantization schemes, Fast Lattice Vector Quantization (FLVQ), Pyramidal Vector Quantization (PVQ) and Scalar Quantization (SQ) are studied in the framework of audio codec G.719. FLVQ is composed of an RE8 -based low-rate lattice vector quantizer and a D8 -based high-rate lattice vector quantizer. PVQ uses pyramidal points in multi-dimensional space and is very suitable for the compression of Laplacian-like sources generated from transform. SQ scheme applies a combination of uniform SQ and entropy coding. Subjective tests of these three versions of audio codecs show that FLVQ and PVQ versions of audio codecs are both better than SQ version for music signals and SQ version of audio codec performs well on speech signals, especially for male speakers. I Acknowledgements I would like to express my sincere gratitude to Ericsson Research, which provides me with such a good thesis work to do. I am indebted to my supervisor, Sebastian Näslund, for sparing time to communicate with me about my work every week and giving me many valuable suggestions. I am also very grateful to Volodya Grancharov and Eric Norvell for their advice and patience as well as consistent encouragement throughout the thesis. My thanks are extended to some other Ericsson researchers for attending the subjective listening evaluation in the thesis. Finally, I want to thank my examiner, Professor Arne Leijon of Royal Institute of Technology (KTH) for reviewing my report very carefully and supporting my work very much. -
MPEG2+4 SDI Codec Manual-V6.Pdf
QVidium™ TECHNOLOGIES, INC. MPEG2+4 SDI IP Codec Model #QVSDI-11-IP User’s Manual v.6 March 24, 2008 Application Firmware Version 21 © 2007-2008 QVidium™ Technologies, Inc. 12989 Chaparral Ridge Road, San Diego, CA 92130 Phone 858.792.6407 • Fax 858.792.9131 User’s Manual v.6 QVidium™ MPEG2+4 SDI IP Codec Table of Contents 1 Introduction.............................................................................................................3 1.1 Overview...........................................................................................................3 1.2 Network Setup .................................................................................................3 1.3 Ping....................................................................................................................5 1.4 Passwords and Security.................................................................................6 1.5 Resetting to Default Settings .........................................................................6 1.6 Upgrading .........................................................................................................6 1.7 System View.....................................................................................................7 2 Encoder Configuration and Operation.............................................................9 2.1 Configuring the Encoder.................................................................................9 2.2 Starting the Encoder .......................................................................................9 -
PXC 550 Wireless Headphones
PXC 550 Wireless headphones Instruction Manual 2 | PXC 550 Contents Contents Important safety instructions ...................................................................................2 The PXC 550 Wireless headphones ...........................................................................4 Package includes ..........................................................................................................6 Product overview .........................................................................................................7 Overview of the headphones .................................................................................... 7 Overview of LED indicators ........................................................................................ 9 Overview of buttons and switches ........................................................................10 Overview of gesture controls ..................................................................................11 Overview of CapTune ................................................................................................12 Getting started ......................................................................................................... 14 Charging basics ..........................................................................................................14 Installing CapTune .....................................................................................................16 Pairing the headphones ...........................................................................................17 -
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. -
(A/V Codecs) REDCODE RAW (.R3D) ARRIRAW
What is a Codec? Codec is a portmanteau of either "Compressor-Decompressor" or "Coder-Decoder," which describes a device or program capable of performing transformations on a data stream or signal. Codecs encode a stream or signal for transmission, storage or encryption and decode it for viewing or editing. Codecs are often used in videoconferencing and streaming media solutions. A video codec converts analog video signals from a video camera into digital signals for transmission. It then converts the digital signals back to analog for display. An audio codec converts analog audio signals from a microphone into digital signals for transmission. It then converts the digital signals back to analog for playing. The raw encoded form of audio and video data is often called essence, to distinguish it from the metadata information that together make up the information content of the stream and any "wrapper" data that is then added to aid access to or improve the robustness of the stream. Most codecs are lossy, in order to get a reasonably small file size. There are lossless codecs as well, but for most purposes the almost imperceptible increase in quality is not worth the considerable increase in data size. The main exception is if the data will undergo more processing in the future, in which case the repeated lossy encoding would damage the eventual quality too much. Many multimedia data streams need to contain both audio and video data, and often some form of metadata that permits synchronization of the audio and video. Each of these three streams may be handled by different programs, processes, or hardware; but for the multimedia data stream to be useful in stored or transmitted form, they must be encapsulated together in a container format. -
Ogg Audio Codec Download
Ogg audio codec download click here to download To obtain the source code, please see the xiph download page. To get set up to listen to Ogg Vorbis music, begin by selecting your operating system above. Check out the latest royalty-free audio codec from Xiph. To obtain the source code, please see the xiph download page. Ogg Vorbis is Vorbis is everywhere! Download music Music sites Donate today. Get Set Up To Listen: Windows. Playback: These DirectShow filters will let you play your Ogg Vorbis files in Windows Media Player, and other OggDropXPd: A graphical encoder for Vorbis. Download Ogg Vorbis Ogg Vorbis is a lossy audio codec which allows you to create and play Ogg Vorbis files using the command-line. The following end-user download links are provided for convenience: The www.doorway.ru DirectShow filters support playing of files encoded with Vorbis, Speex, Ogg Codecs for Windows, version , ; project page - for other. Vorbis Banner Xiph Banner. In our effort to bring Ogg: Media container. This is our native format and the recommended container for all Xiph codecs. Easy, fast, no torrents, no waiting, no surveys, % free, working www.doorway.ru Free Download Ogg Vorbis ACM Codec - A new audio compression codec. Ogg Codecs is a set of encoders and deocoders for Ogg Vorbis, Speex, Theora and FLAC. Once installed you will be able to play Vorbis. Ogg Vorbis MSACM Codec was added to www.doorway.ru by Bjarne (). Type: Freeware. Updated: Audiotags: , 0x Used to play digital music, such as MP3, VQF, AAC, and other digital audio formats. -
Reduced-Complexity End-To-End Variational Autoencoder for on Board Satellite Image Compression
remote sensing Article Reduced-Complexity End-to-End Variational Autoencoder for on Board Satellite Image Compression Vinicius Alves de Oliveira 1,2,* , Marie Chabert 1 , Thomas Oberlin 3 , Charly Poulliat 1, Mickael Bruno 4, Christophe Latry 4, Mikael Carlavan 5, Simon Henrot 5, Frederic Falzon 5 and Roberto Camarero 6 1 IRIT/INP-ENSEEIHT, University of Toulouse, 31071 Toulouse, France; [email protected] (M.C.); [email protected] (C.P.) 2 Telecommunications for Space and Aeronautics (TéSA) Laboratory, 31500 Toulouse, France 3 ISAE-SUPAERO, University of Toulouse, 31055 Toulouse, France; [email protected] 4 CNES, 31400 Toulouse, France; [email protected] (M.B.); [email protected] (C.L.) 5 Thales Alenia Space, 06150 Cannes, France; [email protected] (M.C.); [email protected] (S.H.); [email protected] (F.F.) 6 ESA, 2201 AZ Noordwijk, The Netherlands; [email protected] * Correspondence: [email protected] Abstract: Recently, convolutional neural networks have been successfully applied to lossy image compression. End-to-end optimized autoencoders, possibly variational, are able to dramatically outperform traditional transform coding schemes in terms of rate-distortion trade-off; however, this is at the cost of a higher computational complexity. An intensive training step on huge databases allows autoencoders to learn jointly the image representation and its probability distribution, pos- sibly using a non-parametric density model or a hyperprior auxiliary autoencoder to eliminate the need for prior knowledge. However, in the context of on board satellite compression, time and memory complexities are submitted to strong constraints. -
Opus, a Free, High-Quality Speech and Audio Codec
Opus, a free, high-quality speech and audio codec Jean-Marc Valin, Koen Vos, Timothy B. Terriberry, Gregory Maxwell 29 January 2014 Xiph.Org & Mozilla What is Opus? ● New highly-flexible speech and audio codec – Works for most audio applications ● Completely free – Royalty-free licensing – Open-source implementation ● IETF RFC 6716 (Sep. 2012) Xiph.Org & Mozilla Why a New Audio Codec? http://xkcd.com/927/ http://imgs.xkcd.com/comics/standards.png Xiph.Org & Mozilla Why Should You Care? ● Best-in-class performance within a wide range of bitrates and applications ● Adaptability to varying network conditions ● Will be deployed as part of WebRTC ● No licensing costs ● No incompatible flavours Xiph.Org & Mozilla History ● Jan. 2007: SILK project started at Skype ● Nov. 2007: CELT project started ● Mar. 2009: Skype asks IETF to create a WG ● Feb. 2010: WG created ● Jul. 2010: First prototype of SILK+CELT codec ● Dec 2011: Opus surpasses Vorbis and AAC ● Sep. 2012: Opus becomes RFC 6716 ● Dec. 2013: Version 1.1 of libopus released Xiph.Org & Mozilla Applications and Standards (2010) Application Codec VoIP with PSTN AMR-NB Wideband VoIP/videoconference AMR-WB High-quality videoconference G.719 Low-bitrate music streaming HE-AAC High-quality music streaming AAC-LC Low-delay broadcast AAC-ELD Network music performance Xiph.Org & Mozilla Applications and Standards (2013) Application Codec VoIP with PSTN Opus Wideband VoIP/videoconference Opus High-quality videoconference Opus Low-bitrate music streaming Opus High-quality music streaming Opus Low-delay -
A Multi-Frame PCA-Based Stereo Audio Coding Method
applied sciences Article A Multi-Frame PCA-Based Stereo Audio Coding Method Jing Wang *, Xiaohan Zhao, Xiang Xie and Jingming Kuang School of Information and Electronics, Beijing Institute of Technology, 100081 Beijing, China; [email protected] (X.Z.); [email protected] (X.X.); [email protected] (J.K.) * Correspondence: [email protected]; Tel.: +86-138-1015-0086 Received: 18 April 2018; Accepted: 9 June 2018; Published: 12 June 2018 Abstract: With the increasing demand for high quality audio, stereo audio coding has become more and more important. In this paper, a multi-frame coding method based on Principal Component Analysis (PCA) is proposed for the compression of audio signals, including both mono and stereo signals. The PCA-based method makes the input audio spectral coefficients into eigenvectors of covariance matrices and reduces coding bitrate by grouping such eigenvectors into fewer number of vectors. The multi-frame joint technique makes the PCA-based method more efficient and feasible. This paper also proposes a quantization method that utilizes Pyramid Vector Quantization (PVQ) to quantize the PCA matrices proposed in this paper with few bits. Parametric coding algorithms are also employed with PCA to ensure the high efficiency of the proposed audio codec. Subjective listening tests with Multiple Stimuli with Hidden Reference and Anchor (MUSHRA) have shown that the proposed PCA-based coding method is efficient at processing stereo audio. Keywords: stereo audio coding; Principal Component Analysis (PCA); multi-frame; Pyramid Vector Quantization (PVQ) 1. Introduction The goal of audio coding is to represent audio in digital form with as few bits as possible while maintaining the intelligibility and quality required for particular applications [1]. -
Multi-Core Platforms for Audio and Multimedia Coding Algorithms in Telecommunications
Antti Pakarinen Multi-core platforms for audio and multimedia coding algorithms in telecommunications School of Electrical Engineering Thesis submitted for examination for the degree of Master of Science in Technology. Kirkkonummi 14.9.2012 Thesis supervisor: Prof. Vesa V¨alim¨aki Thesis instructor: M.Sc. (Tech.) Jussi Pekonen Aalto University School of Electrical A! Engineering aalto university abstract of the school of electrical engineering master's thesis Author: Antti Pakarinen Title: Multi-core platforms for audio and multimedia coding algorithms in telecommunications Date: 14.9.2012 Language: English Number of pages: 9+63 Department of Signal processing and Acoustics Professorship: Acoustics and Audio Signal processing Code: S-89 Supervisor: Prof. Vesa V¨alim¨aki Instructor: M.Sc. (Tech.) Jussi Pekonen Multimedia coding algorithms used in telecommunications evolve constantly. Ben- efits and properties of two new hybrid audio codecs (USAC, Opus) were reviewed on a high level as a literature study. It was found that both have succeeded well in subjective sound quality measurements. Tilera TILEPro64-multicore platform and a related software library was evaluated in terms of performance in multimedia coding. The performance in video coding was found to increase with the number of processing cores up to a certain point. With the tested audio codecs, increasing the number of cores did not increase coding performance. Additionally, multicore products of Tilera, Texas Instruments and Freescale were compared. Develop- ment tools of all three vendors were found to have similiar features, despite the differences in hardware architectures. Keywords: Multimedia, audio, video, coding algorithms, multicore aalto-yliopisto diplomityon¨ sahk¨ otekniikan¨ korkeakoulu tiivistelma¨ Tekij¨a:Antti Pakarinen Ty¨onnimi: Moniydinsuorittimet audion ja multimedian koodausalgoritmeille tietoliikenteess¨a P¨aiv¨am¨a¨ar¨a:14.9.2012 Kieli: Englanti Sivum¨a¨ar¨a:9+63 Signaalink¨asittelynja akustiikan laitos Professuuri: Akustiikka ja ¨a¨anenk¨asittely Koodi: S-89 Valvoja: Prof.