Channel Coding for Code Excited Linear Prediction (CELP) Encoded Speech in Mobile Radio Applications
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Digital Speech Processing— Lecture 17
Digital Speech Processing— Lecture 17 Speech Coding Methods Based on Speech Models 1 Waveform Coding versus Block Processing • Waveform coding – sample-by-sample matching of waveforms – coding quality measured using SNR • Source modeling (block processing) – block processing of signal => vector of outputs every block – overlapped blocks Block 1 Block 2 Block 3 2 Model-Based Speech Coding • we’ve carried waveform coding based on optimizing and maximizing SNR about as far as possible – achieved bit rate reductions on the order of 4:1 (i.e., from 128 Kbps PCM to 32 Kbps ADPCM) at the same time achieving toll quality SNR for telephone-bandwidth speech • to lower bit rate further without reducing speech quality, we need to exploit features of the speech production model, including: – source modeling – spectrum modeling – use of codebook methods for coding efficiency • we also need a new way of comparing performance of different waveform and model-based coding methods – an objective measure, like SNR, isn’t an appropriate measure for model- based coders since they operate on blocks of speech and don’t follow the waveform on a sample-by-sample basis – new subjective measures need to be used that measure user-perceived quality, intelligibility, and robustness to multiple factors 3 Topics Covered in this Lecture • Enhancements for ADPCM Coders – pitch prediction – noise shaping • Analysis-by-Synthesis Speech Coders – multipulse linear prediction coder (MPLPC) – code-excited linear prediction (CELP) • Open-Loop Speech Coders – two-state excitation -
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. -
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. -
(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. -
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. -
TMS320C6000 U-Law and A-Law Companding with Software Or The
Application Report SPRA634 - April 2000 TMS320C6000t µ-Law and A-Law Companding with Software or the McBSP Mark A. Castellano Digital Signal Processing Solutions Todd Hiers Rebecca Ma ABSTRACT This document describes how to perform data companding with the TMS320C6000 digital signal processors (DSP). Companding refers to the compression and expansion of transfer data before and after transmission, respectively. The multichannel buffered serial port (McBSP) in the TMS320C6000 supports two companding formats: µ-Law and A-Law. Both companding formats are specified in the CCITT G.711 recommendation [1]. This application report discusses how to use the McBSP to perform data companding in both the µ-Law and A-Law formats. This document also discusses how to perform companding of data not coming into the device but instead located in a buffer in processor memory. Sample McBSP setup codes are provided for reference. In addition, this application report discusses µ-Law and A-Law companding in software. The appendix provides software companding codes and a brief discussion on the companding theory. Contents 1 Design Problem . 2 2 Overview . 2 3 Companding With the McBSP. 3 3.1 µ-Law Companding . 3 3.1.1 McBSP Register Configuration for µ-Law Companding. 4 3.2 A-Law Companding. 4 3.2.1 McBSP Register Configuration for A-Law Companding. 4 3.3 Companding Internal Data. 5 3.3.1 Non-DLB Mode. 5 3.3.2 DLB Mode . 6 3.4 Sample C Functions. 6 4 Companding With Software. 6 5 Conclusion . 7 6 References . 7 TMS320C6000 is a trademark of Texas Instruments Incorporated. -
Bit Rate Adaptation Using Linear Quadratic Optimization for Mobile Video Streaming
applied sciences Article Bit Rate Adaptation Using Linear Quadratic Optimization for Mobile Video Streaming Young-myoung Kang 1 and Yeon-sup Lim 2,* 1 Network Business, Samsung Electronics, Suwon 16677, Korea; [email protected] 2 Department of Convergence Security Engineering, Sungshin Women’s University, Seoul 02844, Korea * Correspondence: [email protected]; Tel.: +82-2-920-7144 Abstract: Video streaming application such as Youtube is one of the most popular mobile applications. To adjust the quality of video for available network bandwidth, a streaming server provides multiple representations of video of which bit rate has different bandwidth requirements. A streaming client utilizes an adaptive bit rate (ABR) scheme to select a proper representation that the network can support. However, in mobile environments, incorrect decisions of an ABR scheme often cause playback stalls that significantly degrade the quality of user experience, which can easily happen due to network dynamics. In this work, we propose a control theory (Linear Quadratic Optimization)- based ABR scheme to enhance the quality of experience in mobile video streaming. Our simulation study shows that our proposed ABR scheme successfully mitigates and shortens playback stalls while preserving the similar quality of streaming video compared to the state-of-the-art ABR schemes. Keywords: dynamic adaptive streaming over HTTP; adaptive bit rate scheme; control theory 1. Introduction Video streaming constitutes a large fraction of current Internet traffic. In particular, video streaming accounts for more than 65% of world-wide mobile downstream traffic [1]. Commercial streaming services such as YouTube and Netflix are implemented based on Citation: Kang, Y.-m.; Lim, Y.-s. -
A Predictive H.263 Bit-Rate Control Scheme Based on Scene
A PRJ3DICTIVE H.263 BIT-RATE CONTROL SCHEME BASED ON SCENE INFORMATION Pohsiang Hsu and K. J. Ray Liu Department of Electrical and Computer Engineering University of Maryland at College Park College Park, Maryland, USA ABSTRACT is typically constant bit rate (e.g. PSTN). Under this cir- cumstance, the encoder’s output bit-rate must be regulated For many real-time visual communication applications such to meet the transmission bandwidth in order to guaran- as video telephony, the transmission channel in used is typ- tee a constant frame rate and delay. Given the channel ically constant bit rate (e.g. PSTN). Under this circum- rate and a target frame rate, we derive the target bits per stance, the encoder’s output bit-rate must be regulated to frame by distribute the channel rate equally to each hme. meet the transmission bandwidth in order to guarantee a Ffate control is typically based on varying the quantization constant frame rate and delay. In this work, we propose a paraneter to meet the target bit budget for each frame. new predictive rate control scheme based on the activity of Many rate control scheme for the block-based DCT/motion the scene to be coded for H.263 encoder compensation video encoders have been proposed in the past. These rate control algorithms are typically focused on MPEG video coding standards. Rate-Distortion based rate 1. INTRODUCTION control for MPEG using Lagrangian optimization [4] have been proposed in the literature. These methods require ex- In the past few years, emerging demands for digital video tensive,rate-distortion measurements and are unsuitable for communication applications such as video conferencing, video real time applications. -
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 ..................................................................................................................................... -
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. -
Lossy Audio Compression Identification
2018 26th European Signal Processing Conference (EUSIPCO) Lossy Audio Compression Identification Bongjun Kim Zafar Rafii Northwestern University Gracenote Evanston, USA Emeryville, USA [email protected] zafar.rafi[email protected] Abstract—We propose a system which can estimate from an compression parameters from an audio signal, based on AAC, audio recording that has previously undergone lossy compression was presented in [3]. The first implementation of that work, the parameters used for the encoding, and therefore identify the based on MP3, was then proposed in [4]. The idea was to corresponding lossy coding format. The system analyzes the audio signal and searches for the compression parameters and framing search for the compression parameters and framing conditions conditions which match those used for the encoding. In particular, which match those used for the encoding, by measuring traces we propose a new metric for measuring traces of compression of compression in the audio signal, which typically correspond which is robust to variations in the audio content and a new to time-frequency coefficients quantized to zero. method for combining the estimates from multiple audio blocks The first work to investigate alterations, such as deletion, in- which can refine the results. We evaluated this system with audio excerpts from songs and movies, compressed into various coding sertion, or substitution, in audio signals which have undergone formats, using different bit rates, and captured digitally as well lossy compression, namely MP3, was presented in [5]. The as through analog transfer. Results showed that our system can idea was to measure traces of compression in the signal along identify the correct format in almost all cases, even at high bit time and detect discontinuities in the estimated framing. -
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..