FLAC Decoding Using GPU Acceleration
Total Page:16
File Type:pdf, Size:1020Kb
Load more
Recommended publications
-
Making Speech Recognition Work on the Web Christopher J. Varenhorst
Making Speech Recognition Work on the Web by Christopher J. Varenhorst Submitted to the Department of Electrical Engineering and Computer Science in partial fulfillment of the requirements for the degree of Masters of Engineering in Computer Science and Engineering at the MASSACHUSETTS INSTITUTE OF TECHNOLOGY May 2011 c Massachusetts Institute of Technology 2011. All rights reserved. Author.................................................................... Department of Electrical Engineering and Computer Science May 20, 2011 Certified by . James R. Glass Principal Research Scientist Thesis Supervisor Certified by . Scott Cyphers Research Scientist Thesis Supervisor Accepted by . Christopher J. Terman Chairman, Department Committee on Graduate Students Making Speech Recognition Work on the Web by Christopher J. Varenhorst Submitted to the Department of Electrical Engineering and Computer Science on May 20, 2011, in partial fulfillment of the requirements for the degree of Masters of Engineering in Computer Science and Engineering Abstract We present an improved Audio Controller for Web-Accessible Multimodal Interface toolkit { a system that provides a simple way for developers to add speech recognition to web pages. Our improved system offers increased usability and performance for users and greater flexibility for developers. Tests performed showed a %36 increase in recognition response time in the best possible networking conditions. Preliminary tests shows a markedly improved users experience. The new Wowza platform also provides a means of upgrading other Audio Controllers easily. Thesis Supervisor: James R. Glass Title: Principal Research Scientist Thesis Supervisor: Scott Cyphers Title: Research Scientist 2 Contents 1 Introduction and Background 7 1.1 WAMI - Web Accessible Multimodal Toolkit . 8 1.1.1 Existing Java applet . 11 1.2 SALT . -
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 -
(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. -
Ease Your Automation, Improve Your Audio, with Ffmpeg
Ease your automation, improve your audio, with FFmpeg A talk by John Warburton, freelance newscaster for the More Radio network, lecturer in the Department of Music and Media in the University of Surrey, Guildford, England Ease your automation, improve your audio, with FFmpeg A talk by John Warburton, who doesn’t have a camera on this machine. My use of Liquidsoap I used it to develop a highly convenient, automated system suitable for in-store radio, sustaining services without time constraints, and targeted music and news services. It’s separate from my professional newscasting job, except I leave it playing when editing my professional on-air bulletins, to keep across world, UK and USA news, and for calming music. In this way, it’s incredibly useful, professionally, right now! What is FFmpeg? It’s not: • “a command line program” • “a tool for pirating” • “a hacker’s plaything” (that’s what they said about GNU/Linux once!) • “out of date” • “full of patent problems” It asks for: • Some technical knowledge of audio-visual containers and codec • Some understanding of what makes up picture and sound files and multiplexes What is FFmpeg? It is: • a world-leading multimedia manipulation framework • a gateway to codecs, filters, multiplexors, demultiplexors, and measuring tools • exceedingly generous at accepting many flavours of audio-visual input • aims to achieve standards-compliant output, and often succeeds • gives the user both library-accessed and command-line-accessed toolkits • is generally among the fastest of its type • incorporates many industry-leading tools • is programmer-friendly • is cross-platform • is open source, by most measures of the term • is at the heart of many broadcast conversion, and signal manipulation systems • is a viable Internet transmission platform Integration with Liquidsoap? (1.4.3) 1. -
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. -
Multimedia Compression Techniques for Streaming
International Journal of Innovative Technology and Exploring Engineering (IJITEE) ISSN: 2278-3075, Volume-8 Issue-12, October 2019 Multimedia Compression Techniques for Streaming Preethal Rao, Krishna Prakasha K, Vasundhara Acharya most of the audio codes like MP3, AAC etc., are lossy as Abstract: With the growing popularity of streaming content, audio files are originally small in size and thus need not have streaming platforms have emerged that offer content in more compression. In lossless technique, the file size will be resolutions of 4k, 2k, HD etc. Some regions of the world face a reduced to the maximum possibility and thus quality might be terrible network reception. Delivering content and a pleasant compromised more when compared to lossless technique. viewing experience to the users of such locations becomes a The popular codecs like MPEG-2, H.264, H.265 etc., make challenge. audio/video streaming at available network speeds is just not feasible for people at those locations. The only way is to use of this. FLAC, ALAC are some audio codecs which use reduce the data footprint of the concerned audio/video without lossy technique for compression of large audio files. The goal compromising the quality. For this purpose, there exists of this paper is to identify existing techniques in audio-video algorithms and techniques that attempt to realize the same. compression for transmission and carry out a comparative Fortunately, the field of compression is an active one when it analysis of the techniques based on certain parameters. The comes to content delivering. With a lot of algorithms in the play, side outcome would be a program that would stream the which one actually delivers content while putting less strain on the audio/video file of our choice while the main outcome is users' network bandwidth? This paper carries out an extensive finding out the compression technique that performs the best analysis of present popular algorithms to come to the conclusion of the best algorithm for streaming data. -
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 -
EMA Mezzanine File Creation Specification and Best Practices Version 1.0.1 For
16530 Ventura Blvd., Suite 400 Encino, CA 91436 818.385.1500 www.entmerch.org EMA Mezzanine File Creation Specification and Best Practices Version 1.0.1 for Digital Audio‐Visual Distribution January 7, 2014 EMA MEZZANINE FILE CREATION SPECIFICATION AND BEST PRACTICES The Mezzanine File Working Group of EMA’s Digital Supply Chain Committee developed the attached recommended Mezzanine File Specification and Best Practices. Why is the Specification and Best Practices document needed? At the request of their customers, content providers and post‐house have been creating mezzanine files unique to each of their retail partners. This causes unnecessary costs in the supply chain and constrains the flow of new content. There is a demand to make more content available for digital distribution more quickly. Sales are lost if content isn’t available to be merchandised. Today’s ecosystem is too manual. Standardization will facilitate automation, reducing costs and increasing speed. Quality control issues slow down today’s processes. Creating one standard mezzanine file instead of many files for the same content should reduce the quantity of errors. And, when an error does occur and is caught by a single customer, it can be corrected for all retailers/distributors. Mezzanine File Working Group Participants in the Mezzanine File Working Group were: Amazon – Ben Waggoner, Ryan Wernet Dish – Timothy Loveridge Google – Bill Kotzman, Doug Stallard Microsoft – Andy Rosen Netflix – Steven Kang , Nick Levin, Chris Fetner Redbox Instant – Joe Ambeault Rovi -
Codec Is a Portmanteau of Either
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. -
Format Support
Episode 6 Format Support FILE FORMAT CODEC Episode Episode Episode Pro EngineCOMMENTS Adaptive bitrate streaming Microsoft Smooth Streaming H.264 (AAC audio) O Windows OS only. Available with Episode Engine License. Apple HLS H.264 (AAC audio) O Available with Episode Engine License. Windows Media WMV, ASF VC-1 O O O WM9 I/O I/O I/O WMV7 and 8 through F4M component on Mac WMA I/O I/O I/O WMA Pro I/O I/O I/O Flash FLV Flash 8 (VP6s/VP6e) I/O I/O I/O SWF Flash 8 (VP6s/VP6e) I/O I/O I/O MOV/MP4/F4V Flash 9 (H.264) I/O I/O I/O F4V as extension to MP4 WebM WebM VP8 O O O Vorbis O O O 3GPP 3GPP AAC I/O I/O I/O H.263 I/O I/O I/O H.264 I/O I/O I/O MainConcept and x264 MPEG-4 I/O I/O I/O 3GPP2 3GPP2 AAC I/O I/O I/O H.263 I/O I/O I/O H.264 I/O I/O I/O MainConcept and x264 MPEG-4 I/O I/O I/O MPEG Elementary Streams MPEG-1 Elementary Stream MPEG-1 (video) I/O I/O I/O MPEG-2 Elementary Stream MPEG-2 I/O I/O I/O MPEG Program Streams PS AAC O O O MainConcept and x264 H.264 I/O I/O I/O MPEG-1/2 (audio) I/O I/O I/O MPEG-2 I/O I/O I/O MPEG-4 I/O I/O I/O MPEG Transport Streams TS AAC I O O AES I I/O I/O H.264 I I/O I/O MainConcept and x264 AVCHD I I I HDV I I/O I/O MPEG - 1/2 (audio) I I/O I/O MPEG - 2 I I/O I/O MPEG - 4 I I/O I/O PCM I I I Matrox MAX H.264 I/O I/O I/O QT codec (*output possible via QT), Requires Matrox MAX hardware - Mac OS X only MPEG System Streams M1A MPEG-1 (audio) I/O I/O I/O M1V MPEG-1 (audio) I/O I/O I/O Episode 6 Format Support Format Support FILE FORMAT CODEC Episode Episode Episode Pro EngineCOMMENTS MPEG-4 MP4 AAC I/O I/O I/O -
Shall I Use Heterogeneous Data Centers? – a Case Study on Video on Demand Systems
Shall I Use Heterogeneous Data Centers? – A Case Study on Video on Demand Systems Shi Feng∗, Hong Zhang∗ and Wenguang Cheny ∗ Department of Computer Science and Technology Tsinghua National Laboratory for Information Science and Technology Beijing, 100084, China Email: ffredfsh,[email protected] y Research Institute of Tsinghua University in Shenzhen Shenzhen, 518057, China Email: [email protected] Abstract—Recent research work and industrial experience In this paper we perform a case study on VOD systems. show that heterogeneous data centers can be more cost- Previous work suggests power and cost efficiency of a plat- effective for some applications, especially those with intensive form should be determined by actually running the workload I/O operations. However, for a given application, it remains difficult to decide whether the application should be deployed on the platform [4], [5]. We conduct extensive experiments on homogeneous nodes or heterogeneous nodes. to find the optimized configuration for VOD systems. We use In this paper, we perform a case study on finding opti- Helix Server with Helix Producer in Windows, and choose mized configurations for Video on Demand(VOD) systems. We VLC+FFmpeg in Linux. We measure the power and cost examine two major modules in VOD systems, the streaming of platform combinations over three types of processors module and the transcoding module. In this study, we measure performance and power on different configurations by varying and three kinds of disks. Each VOD system contains two these factors: processor type, disk type, number of disks, modules. We explore the effectiveness of mapping distinct operating systems and module mapping modes. -
Ffmpeg Codecs Documentation Table of Contents
FFmpeg Codecs Documentation Table of Contents 1 Description 2 Codec Options 3 Decoders 4 Video Decoders 4.1 hevc 4.2 rawvideo 4.2.1 Options 5 Audio Decoders 5.1 ac3 5.1.1 AC-3 Decoder Options 5.2 flac 5.2.1 FLAC Decoder options 5.3 ffwavesynth 5.4 libcelt 5.5 libgsm 5.6 libilbc 5.6.1 Options 5.7 libopencore-amrnb 5.8 libopencore-amrwb 5.9 libopus 6 Subtitles Decoders 6.1 dvbsub 6.1.1 Options 6.2 dvdsub 6.2.1 Options 6.3 libzvbi-teletext 6.3.1 Options 7 Encoders 8 Audio Encoders 8.1 aac 8.1.1 Options 8.2 ac3 and ac3_fixed 8.2.1 AC-3 Metadata 8.2.1.1 Metadata Control Options 8.2.1.2 Downmix Levels 8.2.1.3 Audio Production Information 8.2.1.4 Other Metadata Options 8.2.2 Extended Bitstream Information 8.2.2.1 Extended Bitstream Information - Part 1 8.2.2.2 Extended Bitstream Information - Part 2 8.2.3 Other AC-3 Encoding Options 8.2.4 Floating-Point-Only AC-3 Encoding Options 8.3 flac 8.3.1 Options 8.4 opus 8.4.1 Options 8.5 libfdk_aac 8.5.1 Options 8.5.2 Examples 8.6 libmp3lame 8.6.1 Options 8.7 libopencore-amrnb 8.7.1 Options 8.8 libopus 8.8.1 Option Mapping 8.9 libshine 8.9.1 Options 8.10 libtwolame 8.10.1 Options 8.11 libvo-amrwbenc 8.11.1 Options 8.12 libvorbis 8.12.1 Options 8.13 libwavpack 8.13.1 Options 8.14 mjpeg 8.14.1 Options 8.15 wavpack 8.15.1 Options 8.15.1.1 Shared options 8.15.1.2 Private options 9 Video Encoders 9.1 Hap 9.1.1 Options 9.2 jpeg2000 9.2.1 Options 9.3 libkvazaar 9.3.1 Options 9.4 libopenh264 9.4.1 Options 9.5 libtheora 9.5.1 Options 9.5.2 Examples 9.6 libvpx 9.6.1 Options 9.7 libwebp 9.7.1 Pixel Format 9.7.2 Options 9.8 libx264, libx264rgb 9.8.1 Supported Pixel Formats 9.8.2 Options 9.9 libx265 9.9.1 Options 9.10 libxvid 9.10.1 Options 9.11 mpeg2 9.11.1 Options 9.12 png 9.12.1 Private options 9.13 ProRes 9.13.1 Private Options for prores-ks 9.13.2 Speed considerations 9.14 QSV encoders 9.15 snow 9.15.1 Options 9.16 vc2 9.16.1 Options 10 Subtitles Encoders 10.1 dvdsub 10.1.1 Options 11 See Also 12 Authors 1 Description# TOC This document describes the codecs (decoders and encoders) provided by the libavcodec library.