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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. -
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 ......................................................................................................... -
Lossy Audio Compression Identification
2018 26th European Signal Processing Conference (EUSIPCO) Lossy Audio Compression Identification Bongjun Kim Zafar Rafii Northwestern University Gracenote Evanston, USA Emeryville, USA [email protected] zafar.rafi[email protected] Abstract—We propose a system which can estimate from an compression parameters from an audio signal, based on AAC, audio recording that has previously undergone lossy compression was presented in [3]. The first implementation of that work, the parameters used for the encoding, and therefore identify the based on MP3, was then proposed in [4]. The idea was to corresponding lossy coding format. The system analyzes the audio signal and searches for the compression parameters and framing search for the compression parameters and framing conditions conditions which match those used for the encoding. In particular, which match those used for the encoding, by measuring traces we propose a new metric for measuring traces of compression of compression in the audio signal, which typically correspond which is robust to variations in the audio content and a new to time-frequency coefficients quantized to zero. method for combining the estimates from multiple audio blocks The first work to investigate alterations, such as deletion, in- which can refine the results. We evaluated this system with audio excerpts from songs and movies, compressed into various coding sertion, or substitution, in audio signals which have undergone formats, using different bit rates, and captured digitally as well lossy compression, namely MP3, was presented in [5]. The as through analog transfer. Results showed that our system can idea was to measure traces of compression in the signal along identify the correct format in almost all cases, even at high bit time and detect discontinuities in the estimated framing. -
Hi3716m V430 Hi3716m V430 Cable HD Chip Brief Data Sheet
Hi3716M V430 Hi3716M V430 Cable HD Chip Brief Data Sheet Key Specifications CPU Graphics and Display Processing (Imprex 2.0 High-performance ARM Cortex-A7 processor Processing Engine) Built-in I-cache, D-cache, and L2 cache Hardware TDE Hardware Java acceleration 3-layer OSD Floating-point coprocessor Two video layers Memory Control Interfaces 16-bit and 32-bit color depth DDR3/DDR3L interface Full-hardware anti-aliasing and anti-flicker − Maximum capacity of 512 MB for an external DDR IE, NR, and CCS SDRAM or of 128 MB/256 MB for a built-in DDR DEI SDRAM Audio and Video Interfaces − 16-bit data width PAL, NTSC, and SECAM standard outputs, and forcible A SPI NOR flash/SPI NAND flash, a parallel SLC NAND standard conversion flash, or a SPI NOR flash+a parallel SLC NAND flash Aspect ratio of 4:3 or 16:9, forcible aspect ratio Video Decoding (HiVXE 2.0 Processing Engine) conversion, and free scaling H.265 Main/Main 10@Level 4.1 high-tier 1080p50(60)/1080i/720p/576p/576i/480p/480i outputs H.264 BP/MP/HP@Level 4.2; MVC HD and SD output for the same source MPEG-1 Color gamut compliant with the xvYCC (IEC 61966-2-4) MPEG-2 SP@ML and MP@HL standard MPEG-4 SP@Levels 0–3, ASP@Levels 0–5, GMC, and HDMI 1.4b with HDCP1.4 MPEG-4 short header format (H.263 baseline) Analog video interfaces AVS baseline@Level 6.0 and AVS+ − One CVBS interface VC-1 SP@ML, MP@HL, and AP@Levels 0–3 − One built-in VDAC VP6/VP8 − VBI 1-channel 1080p@60 fps decoding Audio interface − Audio-left and audio-right channels Image Decoding − S/PDIF -
CT-Aacplus — a State-Of-The-Art Audio Coding Scheme
AUDIO CODING CT-aacPlus — a state-of-the-art Audio coding scheme Martin Dietz and Stefan Meltzer Coding Technologies, Germany CT-aacPlus is a combination of Spectral Band Replication (SBR) technology – a bandwidth-extension tool developed by Coding Technologies (CT) in Germany – with the MPEG Advanced Audio Coding (AAC) technology which, to date, has been one of the most efficient traditional perceptual audio-coding schemes. CT-aacPlus is able to deliver high-quality audio signals at bit-rates down to 24 kbit/s for mono and 48 kbit/s for stereo signals. The forthcoming Digital Radio Mondiale (DRM) broadcasting system, among others, will use CT-aacPlus for its audio-coding scheme. CT-aacPlus will enable DRM to deliver an audio quality, in the frequency range below 30 MHz, that is equivalent to – or even better than – that offered by today’s analogue FM services. This article describes the principles of traditional audio coders – and their limitations when used for low bit-rate applications. The second part describes the basic idea of SBR technology and demonstrates the improvements achieved through the combination of SBR technology with traditional audio coders such as AAC and MP3. Advanced Audio Coding (AAC) has so far been one of the most efficient traditional perceptual audio-coding algorithms. In combination with the bandwidth-extension technology, Spectral Band Replication (SBR), the coding efficiency of AAC can be even further improved by at least 30%, thus providing the same audio quality at a 30% lower bit-rate. The combination of AAC and SBR – referred to as CT-aacPlus – will be used by the Digital Radio Mondiale transmission system [1] in the frequency bands below 30 MHz and will provide near- FM sound quality at bit-rates of around 20 kbit/s per audio channel. -
Computational Complexity Optimization on H.264 Scalable/Multiview Video Coding
Computational Complexity Optimization on H.264 Scalable/Multiview Video Coding By Guangyao Zhang A thesis submitted in partial fulfilment for the requirements for the degree of PhD, at the University of Central Lancashire April 2014 Student Declaration Concurrent registration for two or more academic awards I declare that while registered as a candidate for the research degree, I have not been a registered candidate or enrolled student for another award of the University or other academic or professional institution. Material submitted for another award I declare that no material contained in the thesis has been used in any other submission for an academic award and is solely my own work. Collaboration This work presented in this thesis was carried out at the ADSIP (Applied Digital Signal and Image Processing) Research Centre, University of Central Lancashire. The work described in the thesis is entirely the candidate’s own work. Signature of Candidate ________________________________________ Type of Award Doctor of Philosophy School School of Computing, Engineering and Physical Sciences Abstract Abstract The H.264/MPEG-4 Advanced Video Coding (AVC) standard is a high efficiency and flexible video coding standard compared to previous standards. The high efficiency is achieved by utilizing a comprehensive full search motion estimation method. Although the H.264 standard improves the visual quality at low bitrates, it enormously increases the computational complexity. The research described in this thesis focuses on optimization of the computational complexity on H.264 scalable and multiview video coding. Nowadays, video application areas range from multimedia messaging and mobile to high definition television, and they use different type of transmission systems. -
Critical Assessment of Advanced Coding Standards for Lossless Audio Compression
TONNY HIDAYAT et al: A CRITICAL ASSESSMENT OF ADVANCED CODING STANDARDS FOR LOSSLESS .. A Critical Assessment of Advanced Coding Standards for Lossless Audio Compression Tonny Hidayat Mohd Hafiz Zakaria, Ahmad Naim Che Pee Department of Information Technology Faculty of Information and Communication Technology Universitas Amikom Yogyakarta Universiti Teknikal Malaysia Melaka Yogyakarta, Indonesia Melaka, Malaysia [email protected] [email protected], [email protected] Abstract - Image data, text, video, and audio data all require compression for storage issues and real-time access via computer networks. Audio data cannot use compression technique for generic data. The use of algorithms leads to poor sound quality, small compression ratios and algorithms are not designed for real-time access. Lossless audio compression has achieved observation as a research topic and business field of the importance of the need to store data with excellent condition and larger storage charges. This article will discuss and analyze the various lossless and standardized audio coding algorithms that concern about LPC definitely due to its reputation and resistance to compression that is audio. However, another expectation plans are likewise broke down for relative materials. Comprehension of LPC improvements, for example, LSP deterioration procedures is additionally examined in this paper. Keywords - component; Audio; Lossless; Compression; coding. I. INTRODUCTION Compression is to shrink / compress the size. Data compression is a technique to minimize the data so that files can be obtained with a size smaller than the original file size. Compression is needed to minimize the data storage (because the data size is smaller than the original), accelerate information transmission, and limit bandwidth prerequisites. -
Etsi Ts 101 154 V2.4.1 (2018-02)
ETSI TS 101 154 V2.4.1 (2018-02) TECHNICAL SPECIFICATION Digital Video Broadcasting (DVB); Specification for the use of Video and Audio Coding in Broadcast and Broadband Applications 2 ETSI TS 101 154 V2.4.1 (2018-02) Reference RTS/JTC-DVB-377 Keywords broadcasting, digital, DVB, MPEG, TV, UHDTV, video 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 https://portal.etsi.org/TB/ETSIDeliverableStatus.aspx 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. -
Video - Dive Into HTML5
Video - Dive Into HTML5 You are here: Home ‣ Dive Into HTML5 ‣ Video on the Web ❧ Diving In nyone who has visited YouTube.com in the past four years knows that you can embed video in a web page. But prior to HTML5, there was no standards- based way to do this. Virtually all the video you’ve ever watched “on the web” has been funneled through a third-party plugin — maybe QuickTime, maybe RealPlayer, maybe Flash. (YouTube uses Flash.) These plugins integrate with your browser well enough that you may not even be aware that you’re using them. That is, until you try to watch a video on a platform that doesn’t support that plugin. HTML5 defines a standard way to embed video in a web page, using a <video> element. Support for the <video> element is still evolving, which is a polite way of saying it doesn’t work yet. At least, it doesn’t work everywhere. But don’t despair! There are alternatives and fallbacks and options galore. <video> element support IE Firefox Safari Chrome Opera iPhone Android 9.0+ 3.5+ 3.0+ 3.0+ 10.5+ 1.0+ 2.0+ But support for the <video> element itself is really only a small part of the story. Before we can talk about HTML5 video, you first need to understand a little about video itself. (If you know about video already, you can skip ahead to What Works on the Web.) ❧ http://diveintohtml5.org/video.html (1 of 50) [6/8/2011 6:36:23 PM] Video - Dive Into HTML5 Video Containers You may think of video files as “AVI files” or “MP4 files.” In reality, “AVI” and “MP4″ are just container formats. -
White Paper: the AAC Audio Coding Family for Broadcast and Cable TV
WHITE PAPER Fraunhofer Institute for THE AAC AUDIO CODING FAMILY FOR Integrated Circuits IIS BROADCAST AND CABLE TV Management of the institute Prof. Dr.-Ing. Albert Heuberger (executive) Over the last few years, the AAC audio codec family has played an increasingly important Dr.-Ing. Bernhard Grill role as an enabling technology for state-of-the-art multimedia systems. The codecs Am Wolfsmantel 33 combine high audio quality with very low bit-rates, allowing for an impressive audio 91058 Erlangen experience even over channels with limited bandwidth, such as those in broadcasting or www.iis.fraunhofer.de mobile multimedia streaming. The AAC audio coding family also includes special low delay versions, which allow high quality two way communication. Contact Matthias Rose Phone +49 9131 776-6175 [email protected] Contact USA Fraunhofer USA, Inc. Digital Media Technologies* Phone +1 408 573 9900 [email protected] Contact China Toni Fiedler [email protected] Contact Japan Fahim Nawabi Phone: +81 90-4077-7609 [email protected] Contact Korea Youngju Ju Phone: +82 2 948 1291 [email protected] * Fraunhofer USA Digital Media Technologies, a division of Fraunhofer USA, Inc., promotes and supports the products of Fraunhofer IIS in the U. S. Audio and Media Technologies The AAC Audio Coding Family for Broadcast and Cable TV www.iis.fraunhofer.de/audio 1 / 12 A BRIEF HISTORY AND OVERVIEW OF THE MPEG ADVANCED AUDIO CODING FAMILY The first version of Advanced Audio Coding (AAC) was standardized in 1994 as part of the MPEG-2 standard. -
Audio Decoding on the C54X
Audio Decoding on the C54X Alec Robinson, Chuck Lueck, Jon Rowlands AbstractFueled by the excitement over music distribution on the Internet, audio decompression has become a popular topic. There are several different algorithms available, which makes the programmable DSP a nice choice for systems supporting multiple formats. This paper discusses our efforts in porting two audio compression algorithms, MPEG-1 Layer 3 (MP3) and MPEG-2 Advanced Audio Coding (AAC), to the fixed-point C54X DSP. Introduction Within the last year, the popularity of downloadable, compressed audio formats via the internet has skyrocketed. This paper discusses our efforts in porting the decoders of two such compressed audio formats, MPEG-1 Layer 3 (MP3) and MPEG-2 Advanced Audio Coding (AAC), to the C54X fixed-point DSP. MPEG-1 Layer 3 is currently perhaps the most popular compressed audio format, while MPEG-2 AAC offers better audio quality at the same compression ratios. Platform Description The target processor for development was the C54X, TI’s low-power, 16-bit fixed-point DSP. Our initial design goal was to have both decoders running on the C5410 processor, which has 64 kwords of RAM available and up to 100 MIPS of computation power, while our ultimate goal is to get the same functionality running on the C5409, which has 32 kwords of RAM along with a 16 kword ROM for holding tables, constants, etc. For code development, we made significant use of the Spectrum Digital C54X EVM. With this, it was possible to profile the software, perform functional and diagnostic testing on the C54X, etc. -
Hi3796m V200 Brief Data Sheet
Hi3796M V200 Hi3796M V200 Brief Data Sheet Key Specifications High-Performance CPU AAC-LC and HE-AAC V1/V2 decoding 64-bit quad-core high-performance ARM Cortex A53 APE, FLAC, Ogg, AMR-NB, and AMR-WB decoding Integrated multimedia acceleration engine NEON G.711 (u/a) audio decoding Hardware Java acceleration G.711 (u/a), AMR-NB, AMR-WB, and AAC-LC audio Integrated hardware floating-point coprocessor encoding Integrated L1 and L2 Cache HE-AAC transcoding DD (AC3) 3D GPU Channel Decoding and TS De-multiplexing Integrated high-performance multi-core GPU Mali 450 One embedded QAM module, supporting the ITU OpenGL ES 2.0/1.1 and OpenVG 1.1 J83-A/B/C standard Memory Control Interfaces Multi TS processing simultaneously DDR3/DDR3L/DDR4 interface, supporting maximum TS interface for Full Band Capture device 32-bit data width Various descrambling algorithms of the DVB eMMC 5.0 flash interface Security Processing SPI NOR flash interface Advanced CA SPI NAND flash interface Downloadable Conditional Access System Asynchronous/Synchronous NAND flash interface TVOS security mechanism − SLC/MLC flash memory Secure boot, secure storage, and secure upgrade − Maximum 64-bit ECC DRM and hardware watermark Video Decoding (HiVXE 2.0 Processing Engine) HDCP 2.2/1.4 protection for HDMI outputs AVS2 Main-10bit Profile, Level 8.2.60, supporting Graphics and Display Processing (Imprex 2.0 maximum 4K x 2K@60 fps 10-bit decoding Processing Engine) H.265/HEVC Main/Main 10 Profile@Level 5.1 high-tier, Dolby Vision, HDR10, HLG, and SLF