Optimizing Apples Lossless Audio Codec Algorithm Using NVIDIA CUDA
Total Page:16
File Type:pdf, Size:1020Kb
Load more
Recommended publications
-
Lossless Audio Codec Comparison
Contents Introduction 3 1 CD-audio test 4 1.1 CD's used . .4 1.2 Results all CD's together . .4 1.3 Interesting quirks . .7 1.3.1 Mono encoded as stereo (Dan Browns Angels and Demons) . .7 1.3.2 Compressibility . .9 1.4 Convergence of the results . 10 2 High-resolution audio 13 2.1 Nine Inch Nails' The Slip . 13 2.2 Howard Shore's soundtrack for The Lord of the Rings: The Return of the King . 16 2.3 Wasted bits . 18 3 Multichannel audio 20 3.1 Howard Shore's soundtrack for The Lord of the Rings: The Return of the King . 20 A Motivation for choosing these CDs 23 B Test setup 27 B.1 Scripting and graphing . 27 B.2 Codecs and parameters used . 27 B.3 MD5 checksumming . 28 C Revision history 30 Bibliography 31 2 Introduction While testing the efficiency of lossy codecs can be quite cumbersome (as results differ for each person), comparing lossless codecs is much easier. As the last well documented and comprehensive test available on the internet has been a few years ago, I thought it would be a good idea to update. Beside comparing with CD-audio (which is often done to assess codec performance) and spitting out a grand total, this comparison also looks at extremes that occurred during the test and takes a look at 'high-resolution audio' and multichannel/surround audio. While the comparison was made to update the comparison-page on the FLAC website, it aims to be fair and unbiased. -
Advanced Compression and Latency Reduction Techniques Over Data Networks
Western University Scholarship@Western Electronic Thesis and Dissertation Repository 4-22-2015 12:00 AM Advanced Compression and Latency Reduction Techniques Over Data Networks Fuad Shamieh The University of Western Ontario Supervisor Dr. Xianbin Wang The University of Western Ontario Graduate Program in Electrical and Computer Engineering A thesis submitted in partial fulfillment of the equirr ements for the degree in Master of Engineering Science © Fuad Shamieh 2015 Follow this and additional works at: https://ir.lib.uwo.ca/etd Part of the Digital Communications and Networking Commons Recommended Citation Shamieh, Fuad, "Advanced Compression and Latency Reduction Techniques Over Data Networks" (2015). Electronic Thesis and Dissertation Repository. 2844. https://ir.lib.uwo.ca/etd/2844 This Dissertation/Thesis is brought to you for free and open access by Scholarship@Western. It has been accepted for inclusion in Electronic Thesis and Dissertation Repository by an authorized administrator of Scholarship@Western. For more information, please contact [email protected]. ADVANCED COMPRESSION AND LATENCY REDUCTION TECHNIQUES OVER DATA NETWORKS (Thesis format: Monograph) by Fuad Shamieh Graduate Program in Electrical and Computer Engineering A thesis submitted in partial fulfillment of the requirements for the degree of Masters of Engineering Science The School of Graduate and Postdoctoral Studies The University of Western Ontario London, Ontario, Canada c Fuad Shamieh 2015 Abstract Applications and services operating over Internet protocol (IP) networks often suffer from high latency and packet loss rates. These problems are attributed to data congestion resulting from the lack of network resources available to support the demand. The usage of IP networks is not only increasing, but very dynamic as well. -
Download Media Player Codec Pack Version 4.1 Media Player Codec Pack
download media player codec pack version 4.1 Media Player Codec Pack. Description: In Microsoft Windows 10 it is not possible to set all file associations using an installer. Microsoft chose to block changes of file associations with the introduction of their Zune players. Third party codecs are also blocked in some instances, preventing some files from playing in the Zune players. A simple workaround for this problem is to switch playback of video and music files to Windows Media Player manually. In start menu click on the "Settings". In the "Windows Settings" window click on "System". On the "System" pane click on "Default apps". On the "Choose default applications" pane click on "Films & TV" under "Video Player". On the "Choose an application" pop up menu click on "Windows Media Player" to set Windows Media Player as the default player for video files. Footnote: The same method can be used to apply file associations for music, by simply clicking on "Groove Music" under "Media Player" instead of changing Video Player in step 4. Media Player Codec Pack Plus. Codec's Explained: A codec is a piece of software on either a device or computer capable of encoding and/or decoding video and/or audio data from files, streams and broadcasts. The word Codec is a portmanteau of ' co mpressor- dec ompressor' Compression types that you will be able to play include: x264 | x265 | h.265 | HEVC | 10bit x265 | 10bit x264 | AVCHD | AVC DivX | XviD | MP4 | MPEG4 | MPEG2 and many more. File types you will be able to play include: .bdmv | .evo | .hevc | .mkv | .avi | .flv | .webm | .mp4 | .m4v | .m4a | .ts | .ogm .ac3 | .dts | .alac | .flac | .ape | .aac | .ogg | .ofr | .mpc | .3gp and many more. -
A Forensic Database for Digital Audio, Video, and Image Media
THE “DENVER MULTIMEDIA DATABASE”: A FORENSIC DATABASE FOR DIGITAL AUDIO, VIDEO, AND IMAGE MEDIA by CRAIG ANDREW JANSON B.A., University of Richmond, 1997 A thesis submitted to the Faculty of the Graduate School of the University of Colorado Denver in partial fulfillment of the requirements for the degree of Master of Science Recording Arts Program 2019 This thesis for the Master of Science degree by Craig Andrew Janson has been approved for the Recording Arts Program by Catalin Grigoras, Chair Jeff M. Smith Cole Whitecotton Date: May 18, 2019 ii Janson, Craig Andrew (M.S., Recording Arts Program) The “Denver Multimedia Database”: A Forensic Database for Digital Audio, Video, and Image Media Thesis directed by Associate Professor Catalin Grigoras ABSTRACT To date, there have been multiple databases developed for use in many forensic disciplines. There are very large and well-known databases like CODIS (DNA), IAFIS (fingerprints), and IBIS (ballistics). There are databases for paint, shoeprint, glass, and even ink; all of which catalog and maintain information on all the variations of their specific subject matter. Somewhat recently there was introduced the “Dresden Image Database” which is designed to provide a digital image database for forensic study and contains images that are generic in nature, royalty free, and created specifically for this database. However, a central repository is needed for the collection and study of digital audios, videos, and images. This kind of database would permit researchers, students, and investigators to explore the data from various media and various sources, compare an unknown with knowns with the possibility of discovering the likely source of the unknown. -
Ardour Export Redesign
Ardour Export Redesign Thorsten Wilms [email protected] Revision 2 2007-07-17 Table of Contents 1 Introduction 4 4.5 Endianness 8 2 Insights From a Survey 4 4.6 Channel Count 8 2.1 Export When? 4 4.7 Mapping Channels 8 2.2 Channel Count 4 4.8 CD Marker Files 9 2.3 Requested File Types 5 4.9 Trimming 9 2.4 Sample Formats and Rates in Use 5 4.10 Filename Conflicts 9 2.5 Wish List 5 4.11 Peaks 10 2.5.1 More than one format at once 5 4.12 Blocking JACK 10 2.5.2 Files per Track / Bus 5 4.13 Does it have to be a dialog? 10 2.5.3 Optionally store timestamps 5 5 Track Export 11 2.6 General Problems 6 6 MIDI 12 3 Feature Requests 6 7 Steps After Exporting 12 3.1 Multichannel 6 7.1 Normalize 12 3.2 Individual Files 6 7.2 Trim silence 13 3.3 Realtime Export 6 7.3 Encode 13 3.4 Range ad File Export History 7 7.4 Tag 13 3.5 Running a Script 7 7.5 Upload 13 3.6 Export Markers as Text 7 7.6 Burn CD / DVD 13 4 The Current Dialog 7 7.7 Backup / Archiving 14 4.1 Time Span Selection 7 7.8 Authoring 14 4.2 Ranges 7 8 Container Formats 14 4.3 File vs Directory Selection 8 8.1 libsndfile, currently offered for Export 14 4.4 Container Types 8 8.2 libsndfile, also interesting 14 8.3 libsndfile, rather exotic 15 12 Specification 18 8.4 Interesting 15 12.1 Core 18 8.4.1 BWF – Broadcast Wave Format 15 12.2 Layout 18 8.4.2 Matroska 15 12.3 Presets 18 8.5 Problematic 15 12.4 Speed 18 8.6 Not of further interest 15 12.5 Time span 19 8.7 Check (Todo) 15 12.6 CD Marker Files 19 9 Encodings 16 12.7 Mapping 19 9.1 Libsndfile supported 16 12.8 Processing 19 9.2 Interesting 16 12.9 Container and Encodings 19 9.3 Problematic 16 12.10 Target Folder 20 9.4 Not of further interest 16 12.11 Filenames 20 10 Container / Encoding Combinations 17 12.12 Multiplication 20 11 Elements 17 12.13 Left out 21 11.1 Input 17 13 Credits 21 11.2 Output 17 14 Todo 22 1 Introduction 4 1 Introduction 2 Insights From a Survey The basic purpose of Ardour's export functionality is I conducted a quick survey on the Linux Audio Users to create mixdowns of multitrack arrangements. -
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 -
Lossless Audio Codec Comparison
Contents Introduction 3 1 Test setup 4 1.1 Scripting and graphing . .4 1.2 Codecs and parameters used . .5 1.3 WMA, RealAudio and ALAC . .6 2 CD-audio test 8 2.1 CD's used . .8 2.2 Results all CD's together . .9 2.3 Interesting quirks . 12 2.3.1 Mono encoded as stereo (Dan Browns Angels and Demons) 12 2.4 Convergence of the results . 15 3 High-resolution audio 17 3.1 Nine Inch Nails' The Slip . 17 3.2 Howard Shore's soundtrack for The Lord of the Rings: The Re- turn of the King . 20 3.3 Wasted bits . 22 4 Multichannel audio 24 4.1 Howard Shore's soundtrack for The Lord of the Rings: The Re- turn of the King . 24 A Motivation for choosing these CDs 27 Bibliography 31 2 Introduction While testing the efficiency of lossy codecs can be quite cumbersome (as results differ for each person), comparing lossless codecs is much easier. As the last well documented and comprehensive test available on the internet has been a few years ago, I thought it would be a good idea to update. Beside comparing with CD-audio (which is often done to assess codec perfor- mance) and spitting out a grand total, this comparison also looks at extremes that occurred during the test and takes a look at 'high-resolution audio' and multichannel/surround audio. While the comparison was made to update the comparison-page on the FLAC website, it aims to be fair and unbiased. Because of this, you'll probably won't find anything that looks like conclusions: test results are displayed and analysed, but there is no judgement or choice made. -
(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. -
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. -
Data Sheet Rev. D
SigmaDSP Digital Audio Processor with Flexible Audio Routing Matrix Data Sheet ADAU1442/ADAU1445/ADAU1446 FEATURES I2C and SPI control interfaces Fully programmable audio digital signal processor (DSP) for Standalone operation enhanced sound processing Self-boot from serial EEPROM Features SigmaStudio, a proprietary graphical programming 4-channel, 10-bit auxiliary control ADC tool for the development of custom signal flows Multipurpose pins for digital controls and outputs 172 MHz SigmaDSP core; 3584 instructions per sample at 48 kHz Easy implementation of available third-party algorithms 4k parameter RAM, 8k data RAM On-chip regulator for generating 1.8 V from 3.3 V supply Flexible audio routing matrix (FARM) 100-lead TQFP and LQFP packages 24-channel digital input and output Temperature range: −40°C to +105°C Up to 8 stereo asynchronous sample rate converters APPLICATIONS (from 1:8 up to 7.75:1 ratio and 139 dB DNR) Automotive audio processing Stereo S/PDIF input and output Head units Supports serial and TDM I/O, up to fS = 192 kHz Navigation systems Multichannel byte-addressable TDM serial port Rear-seat entertainment systems Pool of 170 ms digital audio delay (at 48 kHz) DSP amplifiers (sound system amplifiers) Clock oscillator for generating master clock from crystal Commercial audio processing PLL for generating core clock from common audio clocks FUNCTIONAL BLOCK DIAGRAM MP[3:0]/ SPI/I2C* SELFBOOT MP[11:4] ADC[3:0] XTALI XTALO ADAU1442/ ADAU1445/ ADAU1446 2 I C/SPI CONTROL MP/ CLOCK INTERFACE AUX ADC PLL OSCILLATOR CLKOUT 1.8V AND SELF-BOOT REGULATOR PROGRAMMABLE AUDIO SPDIFI S/PDIF S/PDIF SPDIFO RECEIVER PROCESSOR CORE TRANSMITTER FLEXIBLE AUDIO ROUTING MATRIX (FARM) SDATA_IN[8:0] SDATA_OUT[8:0] (24-CHANNEL SERIAL DATA SERIAL DATA (24-CHANNEL DIGITAL AUDIO INPUT PORT UP TO 16 CHANNELS OF OUTPUT PORT DIGITAL AUDIO INPUT) (×9) ASYNCHRONOUS (×9) OUTPUT) SAMPLE RATE CONVERTERS BIT CLOCK† BIT CLOCK† (BCLK) SERIAL CLOCK (BCLK) DOMAINS FRAME CLOCK† (×12) FRAME CLOCK† (LRCLK) (LRCLK) *SPI/I2C = THE ADDR0, CLATCH, SCL/CCLK, SDA/COUT, AND ADDR1/CDATA PINS. -
High Bandwidth Acoustics – Transients/Phase/ and the Human Ear
Project Number: FB IQP HB10 HIGH BANDWIDTH ACOUSTICS – TRANSIENTS/PHASE/ AND THE HUMAN EAR An Interactive Qualifying Project Submitted to the Faculty of the WORCESTER POLYTECHNIC INSTITUTE In partial fulfillment of the requirements for the Degree of Bachelor of Science By Samir Zutshi 8/26/2010 Approved: Professor Frederick Bianchi, Major Advisor Daniel Foley, Co-Advisor Abstract Life over 20 kHz has been a debate that resides in the audio society today. The research presented in this paper claims and confirms the existence of data over 20 kHz. Through a series of experiments involving various microphone types, sampling rates and six instruments, a short-term Fourier transform was applied to the transient part of a single note of each instrument. The results are graphically shown in spectrum form alongside the wave file for comparison and analysis. As a result, the experiment has given fundamental evidence that may act as a base for many branches of its application and understanding, primarily being the effect of this lost data on the ears and the mind. Table of Contents 1. List of Figures 3 2. Introduction 5 3. Terminology 5 4. Background Information 11 a. Timeline of Digital Audio 12 5. Preparatory 17 6. Theory 17 7. Zero-Degree Phase Shift Environment 17 8. Short Term Fourier Transform 20 9. Experimental Data 22 10. Analysis and Discussion 34 11. Conclusion 36 12. Bibliography 36 Page | 2 List of Figures 1. Amplitude and Phase diagram 6 2. Harmonic Series 8 3. Reduction of continuous to discrete signal 9 4. Analog Signal 9 5. Resulting Sampled Signal 9 6.