DOCSLIB.ORG

Bachelor Thesis

Total Page：16

File Type：pdf, Size：1020Kb

Download full-text PDF Read full-text
Bachelor Thesis BACHELOR THESIS Perceived Audio Quality in Ogg Vorbis and AAC in Compressed Popular Music in Bit Rates between 96 kbit/s, 128 kbit/s and 192 kbit/s Erica Engborg 2016 Bachelor of Arts Audio Engineering Luleå University of Technology Department of Arts, Communication and Education Perceived Audio Quality in Ogg Vorbis and AAC in Compressed Popular Music in Bit Rates between 96 kbit/s, 128 kbit/s and 192 kbit/s Erica Engborg Bachelor Thesis – Audio Engineering 2016 Supervisor: Jonas Ekeroot Luleå University of Technology Department of Arts, Communication and Education 2 Bachelor Thesis Abstract The audio quality in music has proved to be important by a broad audience. Studies show that 67,1 % of young listeners (Olive, 2011) and 72 % of experienced listeners (Pras et al, 2009) can distinguish lossy coded audio from lossless coded audio. Ogg Vorbis and AAC are two common lossy codecs in music streaming services. Factors that impact the perceived audio quality are genre, bit rate, codec and instrument. An ABX test was performed with 20 trained listeners. The excerpts that were tested were three popular songs from 2015 and 2016 coded in AAC and Ogg Vorbis 96 kbit/s, 128 kbit/s and 192 kbit/s. The results showed that the subjects only could hear a difference between uncoded and lossy coded audio for one excerpt coded in AAC 96 kbit/s. This result does not exactly match those of earlier studies where subjects could hear the difference at higher bit rates, but there are still some similarities. This study is too small to come to any conclusions and more research has to be done on these codecs. The music streaming services that use these codecs already use higher bit rates than these that were tested so the quality is already good enough. 2 Erica Engborg Table of contents Abstract 2 1. Introduction 4 1.1 Background 4 1.1.1 AAC 4 1.1.2 Ogg Vorbis 6 1.2 Previous research 7 1.2.1 Comparison of Vorbis and other codecs in terms of quality 7 1.2.2 Comparison of Vorbis, AAC and other codecs in terms of quality 7 1.2.3 Different sound qualities in different genres 8 1.2.4 Conclusions from earlier research 9 1.3 Purpose 9 2. Method 11 2.1 Stimuli 11 2.1.1 Ogg Vorbis encoding and decoding 12 2.1.2 AAC encoding and decoding 12 2.2 Listening environment 12 2.3 Test subjects 13 2.4 Test design and test instructions 13 3. Results 18 3.1 Statistical analysis 19 3.2 Results analysis 20 4. Discussion 21 5. References 24 Appendix 1 – Spectrograms 26 Appendix 2 – Technical information about the coded songs 33 3 4 Bachelor Thesis 1. Introduction The way we listen and consume music has actually prefer bad sound to good sound”. changed over the last ten years. The With good sound he meant CD quality and compact disc (CD) introduced 44.1/16 to a with bad he meant lossy compressed audio. consumer market. The accessibility of The test result showed that people less than applications on the internet that allow the 20 years old prefer CD-quality to MP3- consumer to buy digital music instead of quality. Similar results have been physical CDs leads to many people using discovered in a similar test with the internet to download and stream music. experienced listeners. (Olive, 2011). Ten years ago limited bandwidth was a problem and audio files with the resolution In Olive’s (2011) test 67,1 % of 44.1 kHz in sample rate and 16 bits led inexperienced teenagers and 72 % in Pras to slow streaming and downloading. To et al’s (2009) test with experienced make music more accessible the music was listeners preferred high quality. Therefore often compressed by an audio codec to it is important to maintain the quality and reduce the file size. In order to reduce the ensure that the lossy compressed audio file size, bits were thrown away. That kind contains as few audible artifacts that is of bit reduction can lead to audible possible to resemble the CD-quality and artifacts. Bit reduction in music is still contain a preferred quality even after the widely used on the Internet in music compression of the audio. streaming services, audio in web TV and DAB-radio. The lossy compression of the Two common codecs that is used for music audio can lead to artifacts and quality streaming and music downloading are Ogg degradation depending on the compression Vorbis and AAC. Ogg Vorbis is used in ratio. The higher the compression ratio is, Spotify, which is a music streaming the more compressed the audio becomes service with 20 million paying users and the more the quality degrades. At the (Karlsson, 2015). AAC is used in iTunes expense of the worse audio quality, this and Apple Music that provides music leads to smaller audio files and faster downloading and streaming for nearly 800 transmission. million accounts (Arora, 2015). Olive (2001) summarized three earlier 1.1 Background studies that “provide some evidence that 1.1.1 AAC listeners generally prefer CD-quality to AAC is a proprietary codec, which means MP3 formats, particularly at lower bit rates that the creators are obligated to guarantee (< 92-128 kbit/s) where the MP3 artifacts a predictable result (Pan, 1995). This become audible”. Olive’s “study was means that the players that intend to have partially motivated by the current popular the compatibility to play AAC need to pay myth – reported widely in the media - that for a license and conform to specifications young consumers are either indifferent to that achieve the intended result. The AAC- the quality of reproduced sound, or they codec is the result of further development 4 Erica Engborg of the MPEG-1 layer III (also known as components of the audio so they MP3) coding from which it uses many could be treated differently. features. The problem with MPEG-1 layer • Model 2 is more complex than III is that it is designed to be backward model 1. It locates the tonal and the nontonal components in the audio. compatible. That results in some To locate these sounds the model compromises that could be improved in a uses data from the two previous new codec, like the way bands are divided. windows to predict the next. The To be backward compatible with MPEG-1 predictability probably makes the layer II and MPEG-1 layer I the audio second model better for must be divided into 32 spectrally equally discriminating the tonal and wide bands on a linear frequency scale. nontonal audio. These bands are then divided in 6 or 18 The models take advantage of the human sub bands depending on the amount of ear’s disability to hear sounds before and transients in each band. In case if many after a transient. Transients have the ability short transients, more sub bands are needed to mask a sound before and after the attack. to hide the bit reduction artifacts (Sayood, The masking is longer after a transient than 2005). before it. Bands that are inaudible for the human ear (that are below the audible SPL- Salomon and Motta (2010) describe the bit level) are removed. The removal of bits in reduction procedure in the following way: bands result in noise that is ideally masked the audio samples are grouped into 32 sub from being below the audible SPL-level. bands by polyphase filters. The audio is The noise can also be masked spectrally by transformed from the time domain to the a tone. If a tone is played, the threshold of frequency domain by using the modified audibility will increase for frequencies discrete cosine transform (MDCT). The around this tone and if another tone is SPL level of the audio for each sub band is played at lower amplitude, spectrally near calculated. Each band is processed the first tone, the second tone will be separately to optimize the quality of the masked. (Salomon & Motta, 2010) lossy compression. A combination of two psychoacoustic models and the number of The noise in the perceptual coding is available bits determines how much noise mainly quantization noise that arises when there will be in each band. The algorithm the audible bands are quantized. This noise determines which psychoacoustic model can be shaped with different noise shaping that is used for each frame. MPEG-1 Layer techniques in the AAC-codec to have less III uses mostly model 2 that is more audible impact on the audio signal. compilation demanding (Salomon & Temporal noise shaping (TNS) shapes the Motta, 2010). Pan (1995) presents the two noise around transients so the masking psychoacoustic models that is used in the effect will cover the noise. This technique MPEG layer III and AAC codecs as: improves the quality especially on voice signals (Salomon & Motta, 2010). The Model 1 identifies and separates • Perceptual noise substitution (PNS) the nontonal and the tonal improves the bit reduction by localizing 5 6 Bachelor Thesis the level of the noise and replaces it with a variable bit rate (VBR). Montnémery and gain value so the noise can be added in the Sandvall (2004) say that VBR can be used decoding. That can result in a lower bit rate to allow the bits to vary over time at a with the same quality (Painter & Spanias, “fixed” bit rate. Each frame does not need 2000). to contain the same amount of bits. With VBR the frames contain the amount of bits 1.1.2 Ogg Vorbis that is needed to avoid audible artifacts in The Vorbis-codec is an open-source codec the best way that is possible with the and doesn’t charge users with any license available amount of bits.
Load more

Recommended publications
  • 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.
    [Show full text]
  • 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.
    [Show full text]
  • 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.
    [Show full text]
  • A Practical Approach to Spatiotemporal Data Compression
    A Practical Approach to Spatiotemporal Data Compres- sion Niall H. Robinson1, Rachel Prudden1 & Alberto Arribas1 1Informatics Lab, Met Office, Exeter, UK. Datasets representing the world around us are becoming ever more unwieldy as data vol- umes grow. This is largely due to increased measurement and modelling resolution, but the problem is often exacerbated when data are stored at spuriously high precisions. In an effort to facilitate analysis of these datasets, computationally intensive calculations are increasingly being performed on specialised remote servers before the reduced data are transferred to the consumer. Due to bandwidth limitations, this often means data are displayed as simple 2D data visualisations, such as scatter plots or images. We present here a novel way to efficiently encode and transmit 4D data fields on-demand so that they can be locally visualised and interrogated. This nascent “4D video” format allows us to more flexibly move the bound- ary between data server and consumer client. However, it has applications beyond purely scientific visualisation, in the transmission of data to virtual and augmented reality. arXiv:1604.03688v2 [cs.MM] 27 Apr 2016 With the rise of high resolution environmental measurements and simulation, extremely large scientific datasets are becoming increasingly ubiquitous. The scientific community is in the pro- cess of learning how to efficiently make use of these unwieldy datasets. Increasingly, people are interacting with this data via relatively thin clients, with data analysis and storage being managed by a remote server. The web browser is emerging as a useful interface which allows intensive 1 operations to be performed on a remote bespoke analysis server, but with the resultant information visualised and interrogated locally on the client1, 2.
    [Show full text]
  • (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.
    [Show full text]
  • Command-Line Sound Editing Wednesday, December 7, 2016
    21m.380 Music and Technology Recording Techniques & Audio Production Workshop: Command-line sound editing Wednesday, December 7, 2016 1 Student presentation (pa1) • 2 Subject evaluation 3 Group picture 4 Why edit sound on the command line? Figure 1. Graphical representation of sound • We are used to editing sound graphically. • But for many operations, we do not actually need to see the waveform! 4.1 Potential applications • • • • • • • • • • • • • • • • 1 of 11 21m.380 · Workshop: Command-line sound editing · Wed, 12/7/2016 4.2 Advantages • No visual belief system (what you hear is what you hear) • Faster (no need to load guis or waveforms) • Efficient batch-processing (applying editing sequence to multiple files) • Self-documenting (simply save an editing sequence to a script) • Imaginative (might give you different ideas of what’s possible) • Way cooler (let’s face it) © 4.3 Software packages On Debian-based gnu/Linux systems (e.g., Ubuntu), install any of the below packages via apt, e.g., sudo apt-get install mplayer. Program .deb package Function mplayer mplayer Play any media file Table 1. Command-line programs for sndfile-info sndfile-programs playing, converting, and editing me- Metadata retrieval dia files sndfile-convert sndfile-programs Bit depth conversion sndfile-resample samplerate-programs Resampling lame lame Mp3 encoder flac flac Flac encoder oggenc vorbis-tools Ogg Vorbis encoder ffmpeg ffmpeg Media conversion tool mencoder mencoder Media conversion tool sox sox Sound editor ecasound ecasound Sound editor 4.4 Real-world
    [Show full text]
  • 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.
    [Show full text]
  • Subjective Assessment of Audio Quality – the Means and Methods Within the EBU
    Subjective assessment of audio quality – the means and methods within the EBU W. Hoeg (Deutsch Telekom Berkom) L. Christensen (Danmarks Radio) R. Walker (BBC) This article presents a number of useful means and methods for the subjective quality assessment of audio programme material in radio and television, developed and verified by EBU Project Group, P/LIST. The methods defined in several new 1. Introduction EBU Recommendations and Technical Documents are suitable for The existing EBU Recommendation, R22 [1], both operational and training states that “the amount of sound programme ma- purposes in broadcasting terial which is exchanged between EBU Members, organizations. and between EBU Members and other production organizations, continues to increase” and that “the only sufficient method of assessing the bal- ance of features which contribute to the quality of An essential prerequisite for ensuring a uniform the sound in a programme is by listening to it.” high quality of sound programmes is to standard- ize the means and methods required for their as- Therefore, “listening” is an integral part of all sessment. The subjective assessment of sound sound and television programme-making opera- quality has for a long time been carried out by tions. Despite the very significant advances of international organizations such as the (former) modern sound monitoring and measurement OIRT [2][3][4], the (former) CCIR (now ITU-R) technology, these essentially objective solutions [5] and the Member organizations of the EBU remain unable to tell us what the programme will itself. It became increasingly important that com- really sound like to the listener at home.
    [Show full text]
  • Equalization
    CHAPTER 3 Equalization 35 The most intuitive effect — OK EQ is EZ U need a Hi EQ IQ MIX SMART QUICK START: Equalization GOALS ■ Fix spectral problems such as rumble, hum and buzz, pops and wind, proximity, and hiss. ■ Fit things into the mix by leveraging of any spectral openings available and through complementary cuts and boosts on spectrally competitive tracks. ■ Feature those aspects of each instrument that players and music fans like most. GEAR ■ Master the user controls for parametric, semiparametric, program, graphic, shelving, high-pass and low-pass filters. ■ Choose the equalizer with the capabilities you need, focusing particularly on the parameters, slopes, and number of bands available. Home stereos have tone controls. We in the studio get equalizers. Perhaps bet- ter described as a “spectral modifier” or “frequency-specific amplitude adjuster,” the equalizer allows the mix engineer to increase or decrease the level of specific frequency ranges within a signal. Having an equalizer is like having multiple volume knobs for a single track. Unlike the volume knob that attenuates or boosts an entire signal, the equalizer is the tool used to turn down or turn up specific frequency portions of any audio track . Out of all signal-processing tools that we use while mixing and tracking, EQ is probably the easiest, most intuitive to use—at first. Advanced applications of equalization, however, demand a deep understanding of the effect and all its Mix Smart. © 2011 Elsevier Inc. All rights reserved. 36 Mix Smart possibilities. Don't underestimate the intellectual challenge and creative poten- tial of this essential mix processor.
    [Show full text]
  • 4. MPEG Layer-3 Audio Encoding
    MPEG Layer-3 An introduction to MPEG Layer-3 K. Brandenburg and H. Popp Fraunhofer Institut für Integrierte Schaltungen (IIS) MPEG Layer-3, otherwise known as MP3, has generated a phenomenal interest among Internet users, or at least among those who want to download highly-compressed digital audio files at near-CD quality. This article provides an introduction to the work of the MPEG group which was, and still is, responsible for bringing this open (i.e. non-proprietary) compression standard to the forefront of Internet audio downloads. 1. Introduction The audio coding scheme MPEG Layer-3 will soon celebrate its 10th birthday, having been standardized in 1991. In its first years, the scheme was mainly used within DSP- based codecs for studio applications, allowing professionals to use ISDN phone lines as cost-effective music links with high sound quality. In 1995, MPEG Layer-3 was selected as the audio format for the digital satellite broadcasting system developed by World- Space. This was its first step into the mass market. Its second step soon followed, due to the use of the Internet for the electronic distribution of music. Here, the proliferation of audio material – coded with MPEG Layer-3 (aka MP3) – has shown an exponential growth since 1995. By early 1999, “.mp3” had become the most popular search term on the Web (according to http://www.searchterms.com). In 1998, the “MPMAN” (by Saehan Information Systems, South Korea) was the first portable MP3 player, pioneering the road for numerous other manufacturers of consumer electronics. “MP3” has been featured in many articles, mostly on the business pages of newspapers and periodicals, due to its enormous impact on the recording industry.
    [Show full text]
  • CALIFORNIA STATE UNIVERSITY, NORTHRIDGE Optimized AV1 Inter
    CALIFORNIA STATE UNIVERSITY, NORTHRIDGE Optimized AV1 Inter Prediction using Binary classification techniques A graduate project submitted in partial fulfillment of the requirements for the degree of Master of Science in Software Engineering by Alex Kit Romero May 2020 The graduate project of Alex Kit Romero is approved: ____________________________________ ____________ Dr. Katya Mkrtchyan Date ____________________________________ ____________ Dr. Kyle Dewey Date ____________________________________ ____________ Dr. John J. Noga, Chair Date California State University, Northridge ii Dedication This project is dedicated to all of the Computer Science professors that I have come in contact with other the years who have inspired and encouraged me to pursue a career in computer science. The words and wisdom of these professors are what pushed me to try harder and accomplish more than I ever thought possible. I would like to give a big thanks to the open source community and my fellow cohort of computer science co-workers for always being there with answers to my numerous questions and inquiries. Without their guidance and expertise, I could not have been successful. Lastly, I would like to thank my friends and family who have supported and uplifted me throughout the years. Thank you for believing in me and always telling me to never give up. iii Table of Contents Signature Page ................................................................................................................................ ii Dedication .....................................................................................................................................
    [Show full text]
  • Data Compression Using Pre-Generated Dictionaries
    Technical Disclosure Commons Defensive Publications Series January 2020 Data compression using pre-generated dictionaries Simon Cooke Follow this and additional works at: https://www.tdcommons.org/dpubs_series Recommended Citation Cooke, Simon, "Data compression using pre-generated dictionaries", Technical Disclosure Commons, (January 16, 2020) https://www.tdcommons.org/dpubs_series/2876 This work is licensed under a Creative Commons Attribution 4.0 License. This Article is brought to you for free and open access by Technical Disclosure Commons. It has been accepted for inclusion in Defensive Publications Series by an authorized administrator of Technical Disclosure Commons. Cooke: Data compression using pre-generated dictionaries Data compression using pre-generated dictionaries ABSTRACT A file is compressed by replacing its characters by codes that are dependent on the statistics of the characters. The character-to-code table, known as the dictionary, is typically incorporated into the compressed file. Popular compression schemes reach theoretical compression limit only asymptotically. Small files or files without much intra-file redundancy, either compress poorly or not at all. This disclosure describes techniques that achieve superior compression, even for small files or files without much intra-file redundancy, by independently maintaining the dictionary at the transmitting and receiving ends of a file transmission, such that the dictionary does not need to be incorporated into the compressed file. KEYWORDS ● Data compression ● Codec dictionary ● Compression dictionary ● Compression ratio ● Page load speed ● Webpage download ● Shannon limit ● Webpage compression ● JavaScript compression ● Web browser Published by Technical Disclosure Commons, 2020 2 Defensive Publications Series, Art. 2876 [2020] BACKGROUND A file is compressed by replacing its characters by codes that are dependent on the statistics of the characters.
    [Show full text]