The MPEG-4 General Audio Coder
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Music 422 Project Report
Music 422 Project Report Jatin Chowdhury, Arda Sahiner, and Abhipray Sahoo October 10, 2020 1 Introduction We present an audio coder that seeks to improve upon traditional coding methods by implementing block switching, spectral band replication, and gain-shape quantization. Block switching improves on coding of transient sounds. Spectral band replication exploits low sensitivity of human hearing towards high frequencies. It fills in any missing coded high frequency content with information from the low frequencies. We chose to do gain-shape quantization in order to explicitly code the energy of each band, preserving the perceptually important spectral envelope more accurately. 2 Implementation 2.1 Block Switching Block switching, as introduced by Edler [1], allows the audio coder to adaptively change the size of the block being coded. Modern audio coders typically encode frequency lines obtained using the Modified Discrete Cosine Transform (MDCT). Using longer blocks for the MDCT allows forbet- ter spectral resolution, while shorter blocks have better time resolution, due to the time-frequency trade-off known as the Fourier uncertainty principle [2]. Due to their poor time resolution, using long MDCT blocks for an audio coder can result in “pre-echo” when the coder encodes transient sounds. Block switching allows the coder to use long MDCT blocks for normal signals being encoded and use shorter blocks for transient parts of the signal. In our coder, we implement the block switching algorithm introduced in [1]. This algorithm consists of four types of block windows: long, short, start, and stop. Long and short block windows are simple sine windows, as defined in [3]: (( ) ) 1 π w[n] = sin n + (1) 2 N where N is the length of the block. -
(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. -
International Organisation for Standardisation Organisation Internationale De Normalisation Iso/Iec Jtc1/Sc29/Wg11 Coding of Moving Pictures and Audio
INTERNATIONAL ORGANISATION FOR STANDARDISATION ORGANISATION INTERNATIONALE DE NORMALISATION ISO/IEC JTC1/SC29/WG11 CODING OF MOVING PICTURES AND AUDIO ISO/IEC JTC1/SC29/WG11 MPEG98/N2425 Atlantic City - October 1998 Source: Audio and Test subgroup Status: Approved Title: MPEG-4 Audio verification test results: Audio on Internet Authors: Eric Scheirer, Sang-Wook Kim, Martin Dietz Table of Contents 1. Introduction.....................................................................................................................................2 2. Test motivation................................................................................................................................2 3. Codecs under Test ...........................................................................................................................3 4. Test Material...................................................................................................................................5 5. Test methodology ............................................................................................................................7 6. Test stimuli .....................................................................................................................................7 7. Test sessions ...................................................................................................................................8 8. Data Analysis..................................................................................................................................8 -
What Is Ogg Vorbis?
Ogg Vorbis Audio Compression Format Norat Rossello Castilla 5/18/2005 Ogg Vorbis 1 What is Ogg Vorbis? z Audio compression format z Comparable to MP3, VQF, AAC, TwinVQ z Free, open and unpatented z Broadcasting, radio station and television by internet ( = Streaming) 5/18/2005 Ogg Vorbis 2 1 About the name… z Ogg = name of Xiph.org container format for audio, video and metadata z Vorbis = name of specific audio compression scheme designed to be contained in Ogg FOR MORE INFO... https://www.xiph.org 5/18/2005 Ogg Vorbis 3 Some comercial characteristics z The official mime type was approved in February 2003 z Posible to encode all music or audio content in Vorbis z Designed to not be proprietary or patented audio format z Patent and licensed-free z Specification in public domain 5/18/2005 Ogg Vorbis 4 2 Audio Compression z Two classes of compression algorithms: - Lossless - Lossy FOR MORE INFO... http://www.firstpr.com.au/audiocomp 5/18/2005 Ogg Vorbis 5 Lossless algorithms z Produce compressed data that can be decoded to output that is identical to the original. z Zip, FLAC for audio 5/18/2005 Ogg Vorbis 6 3 Lossy algorithms z Discard data in order to compress it better than would normally be possible z VORBIS, MP3, JPEG z Throw away parts of the audio waveform that are irrelevant. 5/18/2005 Ogg Vorbis 7 Ogg Vorbis - Compression Factors z Vorbis is an audio codec that generates 16 bit samples at 16KHz to 48KHz, providing variable bit rates from 16 to 128 Kbps per channel FOR MORE INFO.. -
User Manual Contents
Register your product and get support at 8209 www.philips.com/welcome 48PFS8209 48PFS8209 55PFS8209 55PFS8209 User Manual Contents 6.7 Batteries 28 1 TV Tour 4 6.8 Cleaning 28 1.1 Android TV 4 1.2 Apps 4 7 Gesture Control 30 1.3 Movies and Missed Shows 4 7.1 About Gesture Control 30 1.4 Social Networks 4 7.2 Camera 30 1.5 Pause TV and Recordings 4 7.3 Hand Gestures 30 1.6 Gaming 4 7.4 Gesture Overview 30 1.7 Skype 4 7.5 Tips 30 1.8 3D 5 1.9 Smartphones and Tablets 5 8 Home Menu 32 8.1 Open the Home Menu 32 2 Setting Up 6 8.2 Overview 32 2.1 Read Safety 6 8.3 Notifications 32 2.2 TV Stand and Wall Mounting 6 8.4 Search 32 2.3 Tips on Placement 6 2.4 Power Cable 6 9 Now on TV 33 2.5 Antenna Cable 6 9.1 About Now on TV 33 2.6 Satellite Dish 7 9.2 What You Need 33 9.3 Using Now on TV 33 3 Network 8 3.1 Connect to Network 8 10 Apps 34 3.2 Network Settings 9 10.1 About Apps 34 3.3 Network Devices 10 10.2 Install an App 34 3.4 File Sharing 10 10.3 Start an App 34 10.4 Chrome™ 34 4 Connections 11 10.5 App Lock 34 4.1 Tips on Connections 11 10.6 Widgets 35 4.2 EasyLink HDMI CEC 12 10.7 Remove Apps and Widgets 35 4.3 CI+ CAM with Smart Card 13 10.8 Clear Internet Memory 35 4.4 Set-Top Box - STB 14 10.9 Android Settings 35 4.5 Satellite Receiver 14 10.10 Terms of Use - Apps 36 4.6 Home Theatre System - HTS 15 4.7 Blu-ray Disc Player 16 11 Video on Demand 37 4.8 DVD Player 16 11.1 About Video on Demand 37 4.9 Game Console 17 11.2 Rent a Movie 37 4.10 Gamepad 17 11.3 Streaming 37 4.11 USB Hard Drive 18 12 TV on Demand 38 4.12 USB Keyboard or Mouse -
Center for Signal Processing
21 3. LOSSLESS/LOSSY COMPRESSION Overview of Lossless Compression: It is also known as: • Noiseless coding • Lossless coding • Invertible coding • Entropy coding • Data compaction codes. They can perfectly recover original data (if no storage or transmission bit errors, i.e., noiseless channel). 1. They normally have variable length binary codewords to produce variable numbers of bits per symbol. 2. Only works for digital sources. Properties of Variable rate (length) systems: • They can more efficiently trade off quality/distortion (noise) and rate. • They are generally more complex and costly to build. • They require buffering to match fixed-rate transmission system requirements. • They tend to suffer catastrophic transmission error propagation; in other words, once in error then the error grows quickly. 3. But they can provide produce superior rate/distortion (noise) trade-off. For instance, in image compression we can use more bits for edges, fewer for flat areas. Similarly, more bits can be assigned for plosives, fewer for vowels in speech compression. General idea: Code highly probable symbols into short binary sequences, low probability symbols into long binary sequences, so that average is minimized. Most famous examples: • Morse code: Consider dots and dashes as binary levels “0” and “1”. Assign codeword length inversely proportional to letter relative frequencies. In other words, most frequent letter (message) is assigned the smallest number of bits, whereas the least likely message has the longest codeword, such as “e” and “”z” in English alphabet. Total number of bits per second would much less than if we have used fixed-rate ASCII representation for alphanumerical data. • Huffman code 1952, employed in UNIX compact utility and many standards, known to be optimal under specific constraints. -
Overview of the MPEG 4 Standard
).4%2.!4)/.!,/2'!.)3!4)/.&/234!.$!2$)3!4)/. /2'!.)3!4)/.).4%2.!4)/.!,%$%./2-!,)3!4)/. )3/)%#*4#3#7' #/$).'/&-/6).'0)#452%3!.$!5$)/ )3/)%#*4#3#7'. $ECEMBER-AUI 3OURCE 7'-0%' 3TATUS &INAL 4ITLE -0%' /VERVIEW -AUI6ERSION %DITOR 2OB+OENEN /VERVIEWOFTHE-0%' 3TANDARD %XECUTIVE/VERVIEW MPEG-4 is an ISO/IEC standard developed by MPEG (Moving Picture Experts Group), the committee that also developed the Emmy Award winning standards known as MPEG-1 and MPEG-2. These standards made interactive video on CD-ROM and Digital Television possible. MPEG-4 is the result of another international effort involving hundreds of researchers and engineers from all over the world. MPEG-4, whose formal ISO/IEC designation is ISO/IEC 14496, was finalized in October 1998 and became an International Standard in the first months of 1999. The fully backward compatible extensions under the title of MPEG-4 Version 2 were frozen at the end of 1999, to acquire the formal International Standard Status early 2000. Some work, on extensions in specific domains, is still in progress. MPEG-4 builds on the proven success of three fields: • Digital television; • Interactive graphics applications (synthetic content) ; • Interactive multimedia (World Wide Web, distribution of and access to content) MPEG-4 provides the standardized technological elements enabling the integration of the production, distribution and content access paradigms of the three fields. More information about MPEG-4 can be found at MPEG’s home page (case sensitive): http://www.cselt.it/mpeg . This web page contains links to a wealth of information about MPEG, including much about MPEG-4, many publicly available documents, several lists of ‘Frequently Asked Questions’ and links to other MPEG-4 web pages. -
Input Formats & Codecs
Input Formats & Codecs Pivotshare offers upload support to over 99.9% of codecs and container formats. Please note that video container formats are independent codec support. Input Video Container Formats (Independent of codec) 3GP/3GP2 ASF (Windows Media) AVI DNxHD (SMPTE VC-3) DV video Flash Video Matroska MOV (Quicktime) MP4 MPEG-2 TS, MPEG-2 PS, MPEG-1 Ogg PCM VOB (Video Object) WebM Many more... Unsupported Video Codecs Apple Intermediate ProRes 4444 (ProRes 422 Supported) HDV 720p60 Go2Meeting3 (G2M3) Go2Meeting4 (G2M4) ER AAC LD (Error Resiliant, Low-Delay variant of AAC) REDCODE Supported Video Codecs 3ivx 4X Movie Alaris VideoGramPiX Alparysoft lossless codec American Laser Games MM Video AMV Video Apple QuickDraw ASUS V1 ASUS V2 ATI VCR-2 ATI VCR1 Auravision AURA Auravision Aura 2 Autodesk Animator Flic video Autodesk RLE Avid Meridien Uncompressed AVImszh AVIzlib AVS (Audio Video Standard) video Beam Software VB Bethesda VID video Bink video Blackmagic 10-bit Broadway MPEG Capture Codec Brooktree 411 codec Brute Force & Ignorance CamStudio Camtasia Screen Codec Canopus HQ Codec Canopus Lossless Codec CD Graphics video Chinese AVS video (AVS1-P2, JiZhun profile) Cinepak Cirrus Logic AccuPak Creative Labs Video Blaster Webcam Creative YUV (CYUV) Delphine Software International CIN video Deluxe Paint Animation DivX ;-) (MPEG-4) DNxHD (VC3) DV (Digital Video) Feeble Files/ScummVM DXA FFmpeg video codec #1 Flash Screen Video Flash Video (FLV) / Sorenson Spark / Sorenson H.263 Forward Uncompressed Video Codec fox motion video FRAPS: -
Coding and Compression
Chapter 3 Multimedia Systems Technology: Co ding and Compression 3.1 Intro duction In multimedia system design, storage and transport of information play a sig- ni cant role. Multimedia information is inherently voluminous and therefore requires very high storage capacity and very high bandwidth transmission capacity.For instance, the storage for a video frame with 640 480 pixel resolution is 7.3728 million bits, if we assume that 24 bits are used to en- co de the luminance and chrominance comp onents of each pixel. Assuming a frame rate of 30 frames p er second, the entire 7.3728 million bits should b e transferred in 33.3 milliseconds, which is equivalent to a bandwidth of 221.184 million bits p er second. That is, the transp ort of such large number of bits of information and in such a short time requires high bandwidth. Thereare two approaches that arepossible - one to develop technologies to provide higher bandwidth of the order of Gigabits per second or more and the other to nd ways and means by which the number of bits to betransferred can be reduced, without compromising the information content. It amounts to saying that we need a transformation of a string of characters in some representation such as ASCI I into a new string e.g., of bits that contains the same information but whose length must b e as small as p ossible; i.e., data compression. Data compression is often referred to as co ding, whereas co ding is a general term encompassing any sp ecial representation of data that achieves a given goal. -
Experiment 7 IMAGE COMPRESSION I Introduction
Experiment 7 IMAGE COMPRESSION I Introduction A digital image obtained by sampling and quantizing a continuous tone picture requires an enormous storage. For instance, a 24 bit color image with 512x512 pixels will occupy 768 Kbyte storage on a disk, and a picture twice of this size will not fit in a single floppy disk. To transmit such an image over a 28.8 Kbps modem would take almost 4 minutes. The purpose for image compression is to reduce the amount of data required for representing sampled digital images and therefore reduce the cost for storage and transmission. Image compression plays a key role in many important applications, including image database, image communications, remote sensing (the use of satellite imagery for weather and other earth-resource applications), document and medical imaging, facsimile transmission (FAX), and the control of remotely piloted vehicles in military, space, and hazardous waste control applications. In short, an ever-expanding number of applications depend on the efficient manipulation, storage, and transmission of binary, gray-scale, or color images. An important development in image compression is the establishment of the JPEG standard for compression of color pictures. Using the JPEG method, a 24 bit/pixel color images can be reduced to between 1 to 2 bits/pixel, without obvious visual artifacts. Such reduction makes it possible to store and transmit digital imagery with reasonable cost. It also makes it possible to download a color photograph almost in an instant, making electronic publishing/advertising on the Web a reality. Prior to this event, G3 and G4 standards have been developed for compression of facsimile documents, reducing the time for transmitting one page of text from about 6 minutes to 1 minute. -
Forcepoint DLP Supported File Formats and Size Limits
Forcepoint DLP Supported File Formats and Size Limits Supported File Formats and Size Limits | Forcepoint DLP | v8.8.1 This article provides a list of the file formats that can be analyzed by Forcepoint DLP, file formats from which content and meta data can be extracted, and the file size limits for network, endpoint, and discovery functions. See: ● Supported File Formats ● File Size Limits © 2021 Forcepoint LLC Supported File Formats Supported File Formats and Size Limits | Forcepoint DLP | v8.8.1 The following tables lists the file formats supported by Forcepoint DLP. File formats are in alphabetical order by format group. ● Archive For mats, page 3 ● Backup Formats, page 7 ● Business Intelligence (BI) and Analysis Formats, page 8 ● Computer-Aided Design Formats, page 9 ● Cryptography Formats, page 12 ● Database Formats, page 14 ● Desktop publishing formats, page 16 ● eBook/Audio book formats, page 17 ● Executable formats, page 18 ● Font formats, page 20 ● Graphics formats - general, page 21 ● Graphics formats - vector graphics, page 26 ● Library formats, page 29 ● Log formats, page 30 ● Mail formats, page 31 ● Multimedia formats, page 32 ● Object formats, page 37 ● Presentation formats, page 38 ● Project management formats, page 40 ● Spreadsheet formats, page 41 ● Text and markup formats, page 43 ● Word processing formats, page 45 ● Miscellaneous formats, page 53 Supported file formats are added and updated frequently. Key to support tables Symbol Description Y The format is supported N The format is not supported P Partial metadata -
Structural Analysis of Low Latency Audio Coding Schemes
Structural analysis of low latency audio coding schemes Manfred Lutzky, Markus Schnell, Markus Schmidt and Ralf Geiger Fraunhofer Institute for Integrated Circuits IIS, Am Wolfsmantel 33, 91058 Erlangen, Germany Correspondence should be addressed to: [email protected] ABSTRACT Low latency audio coding gains increasing importance among upcoming high quality communication applications like video conferencing and VoIP. This paper provides a comparison of two low latency audio codecs suitable for these tasks: MPEG-4 ER AAC-LD and ITU-T G.722.1 Annex C. Despite their similar coding strategies, both codecs show significant differences with respect to used tools and coding performance. A comparison of the coding tools is provided and the influence on different signal classes is discussed. Highly sophisticated audio codecs are necessary for 1. INTRODUCTION encoding audio data at low bit-rates and to achieve natural sound quality which helps to improve the Current VoIP and video conferencing systems are based intelligibility in difficult environments such as on the usage of no longer up-to-date telephone codecs, understanding fast and inarticulate speakers in a foreign e.g. ITU-T G.711, and low bitrate speech codecs, e.g. language. Application scenarios nowadays no longer ITU-T G.728. Unfortunately, these codecs do not make include only pure speech transmission, but also the the most of the available transmission bandwidth. necessity of a transmission channel for music, e.g. for a Whereas telephone codecs prove to be too expensive multimedia business presentation broadcast over a video with respect to bandwidth, speech codecs are not able to conferencing system.