CELT: a Low-Latency, High-Quality Audio Codec
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C-Based Hardware Design of Imdct Accelerator for Ogg Vorbis Decoder
C-BASED HARDWARE DESIGN OF IMDCT ACCELERATOR FOR OGG VORBIS DECODER Shinichi Maeta1, Atsushi Kosaka1, Akihisa Yamada1, 2, Takao Onoye1, Tohru Chiba1, 2, and Isao Shirakawa1 1Department of Information Systems Engineering, 2Sharp Corporation Graduate School of Information Science and Technology, 2613-1 Ichinomoto, Tenri, Nara, 632-8567 Japan Osaka University phone: +81 743 65 2531, fax: +81 743 65 3963, 2-1 Yamada-oka, Suita, Osaka, 565-0871 Japan email: [email protected], phone: +81 6 6879 7808, fax: +81 6 6875 5902, [email protected] email: {maeta, kosaka, onoye, sirakawa}@ist.osaka-u.ac.jp ABSTRACT ARM7TDMI is used as the embedded processor since it has This paper presents hardware design of an IMDCT accelera- come into wide use recently. tor for an Ogg Vorbis decoder using a C-based design sys- tem. Low power implementation of audio codec is important 2. OGG VORBIS CODEC in order to achieve long battery life of portable audio de- 2.1 Ogg Vorbis Overview vices. Through the computational cost analysis of the whole decoding process, it is found that Ogg Vorbis requires higher Figure 1 shows a block diagram of the Ogg Vorbis codec operation frequency of an embedded processor than MPEG processes outlined below. Audio. In order to reduce the CPU load, an accelerator is designed as specific hardware for IMDCT, which is detected MDCT Psycho Audio Remove Channel Acoustic VQ as the most computation-intensive functional block. Real- Signal Floor Coupling time decoding of Ogg Vorbis is achieved with the accelera- FFT Model Ogg Vorbis tor and an embedded processor both run at 36MHz. -
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 -
Blackberry QNX Multimedia Suite
PRODUCT BRIEF QNX Multimedia Suite The QNX Multimedia Suite is a comprehensive collection of media technology that has evolved over the years to keep pace with the latest media requirements of current-day embedded systems. Proven in tens of millions of automotive infotainment head units, the suite enables media-rich, high-quality playback, encoding and streaming of audio and video content. The multimedia suite comprises a modular, highly-scalable architecture that enables building high value, customized solutions that range from simple media players to networked systems in the car. The suite is optimized to leverage system-on-chip (SoC) video acceleration, in addition to supporting OpenMAX AL, an industry open standard API for application-level access to a device’s audio, video and imaging capabilities. Overview Consumer’s demand for multimedia has fueled an anywhere- o QNX SDK for Smartphone Connectivity (with support for Apple anytime paradigm, making multimedia ubiquitous in embedded CarPlay and Android Auto) systems. More and more embedded applications have require- o Qt distributions for QNX SDP 7 ments for audio, video and communication processing capabilities. For example, an infotainment system’s media player enables o QNX CAR Platform for Infotainment playback of content, stored either on-board or accessed from an • Support for a variety of external media stores external drive, mobile device or streamed over IP via a browser. Increasingly, these systems also have streaming requirements for Features at a Glance distributing content across a network, for instance from a head Multimedia Playback unit to the digital instrument cluster or rear seat entertainment units. Multimedia is also becoming pervasive in other markets, • Software-based audio CODECs such as medical, industrial, and whitegoods where user interfaces • Hardware accelerated video CODECs are increasingly providing users with a rich media experience. -
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. -
Detail Streaming Support Protocols
Encore+ User Guide Detail Streaming Support Protocols Supported Audio Codecs Supported Container Formats • MP3 • WAV • AAC • M4A • FLAC • OGG • LPCM/WAV/AIFF • AIFF • ALAC Supported Protocols • WMA, WMA9 • SHOUTcast • Ogg Vorbis • HTTPS Supported Playlist • WMA streaming • ASX • RTSP/SDP • M3U • PLS • WPL 43 Detail Audio Codec Support Encore+ User Guide Supported MP3 encoding parameters • Sampling rates [kHz]: 32, 44.1, 48 • Resolution [bits]: 16 • Bit rate [kbps]: 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, VBR • Channels: stereo, joined stereo, mono • MP3PRO playback • MP3 File extensions: *.mp3 • Decoding of ID3v1, ID3v2, MP3 ID tags including optional album art in .jpeg format up to 2 megapixels • Gapless MP3: Playback is gapless if the container provides LAME encoder delay and padding tags. Supported Vorbis encoding parameters • Sampling rates [kHz]: 32, 44.1, 48 • Resolution [bits]: 16 • Nominal bit rate [kbps] (quality level): 80 (Q1), 96 (Q2), 112 (Q3), 128 (Q4), 160 (Q5), 192 (Q6), • Channels: stereo • The audio player supports reading of Vorbis content stored in Ogg containers. Supported file name extensions: *.ogg and *.oga. • The audio player supports decoding of Vorbis comments. NOTE: There is no specification for tag names. The system relies on the OSS implementation. • Tag names decoded: TITLE, ALBUM, ARTIST, GENRE. • Binary data (e.g. for album art) is not supported. • The audio player supports gapless Vorbis playback. Supported FLAC encoding parameters • Sampling rates [kHz]: 44.1, 48, 88.2, 96, 176.4, 192 • Resolution [bits]: 16, 24 • Channels: stereo, mono • The audio player supports reading of FLAC content stored in native FLAC containers. -
(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. -
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. -
Fast Computational Structures for an Efficient Implementation of The
ARTICLE IN PRESS Signal Processing ] (]]]]) ]]]–]]] Contents lists available at ScienceDirect Signal Processing journal homepage: www.elsevier.com/locate/sigpro Fast computational structures for an efficient implementation of the complete TDAC analysis/synthesis MDCT/MDST filter banks Vladimir Britanak a,Ã, Huibert J. Lincklaen Arrie¨ns b a Institute of Informatics, Slovak Academy of Sciences, Dubravska cesta 9, 845 07 Bratislava, Slovak Republic b Delft University of Technology, Department of Electrical Engineering, Mathematics and Computer Science, Mekelweg 4, 2628 CD Delft, The Netherlands article info abstract Article history: A new fast computational structure identical both for the forward and backward Received 27 May 2008 modified discrete cosine/sine transform (MDCT/MDST) computation is described. It is Received in revised form the result of a systematic construction of a fast algorithm for an efficient implementa- 8 January 2009 tion of the complete time domain aliasing cancelation (TDAC) analysis/synthesis MDCT/ Accepted 10 January 2009 MDST filter banks. It is shown that the same computational structure can be used both for the encoder and the decoder, thus significantly reducing design time and resources. Keywords: The corresponding generalized signal flow graph is regular and defines new sparse Modified discrete cosine transform matrix factorizations of the discrete cosine transform of type IV (DCT-IV) and MDCT/ Modified discrete sine transform MDST matrices. The identical fast MDCT computational structure provides an efficient Modulated -
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).