Companding Techniques for High Dynamic Range Audio CODEC Receiver Path by Yunjie Ma S.B
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Data Compression: Dictionary-Based Coding 2 / 37 Dictionary-Based Coding Dictionary-Based Coding
Dictionary-based Coding already coded not yet coded search buffer look-ahead buffer cursor (N symbols) (L symbols) We know the past but cannot control it. We control the future but... Last Lecture Last Lecture: Predictive Lossless Coding Predictive Lossless Coding Simple and effective way to exploit dependencies between neighboring symbols / samples Optimal predictor: Conditional mean (requires storage of large tables) Affine and Linear Prediction Simple structure, low-complex implementation possible Optimal prediction parameters are given by solution of Yule-Walker equations Works very well for real signals (e.g., audio, images, ...) Efficient Lossless Coding for Real-World Signals Affine/linear prediction (often: block-adaptive choice of prediction parameters) Entropy coding of prediction errors (e.g., arithmetic coding) Using marginal pmf often already yields good results Can be improved by using conditional pmfs (with simple conditions) Heiko Schwarz (Freie Universität Berlin) — Data Compression: Dictionary-based Coding 2 / 37 Dictionary-based Coding Dictionary-Based Coding Coding of Text Files Very high amount of dependencies Affine prediction does not work (requires linear dependencies) Higher-order conditional coding should work well, but is way to complex (memory) Alternative: Do not code single characters, but words or phrases Example: English Texts Oxford English Dictionary lists less than 230 000 words (including obsolete words) On average, a word contains about 6 characters Average codeword length per character would be limited by 1 -
A Multidimensional Companding Scheme for Source Coding Witha Perceptually Relevant Distortion Measure
A MULTIDIMENSIONAL COMPANDING SCHEME FOR SOURCE CODING WITHA PERCEPTUALLY RELEVANT DISTORTION MEASURE J. Crespo, P. Aguiar R. Heusdens Department of Electrical and Computer Engineering Department of Mediamatics Instituto Superior Técnico Technische Universiteit Delft Lisboa, Portugal Delft, The Netherlands ABSTRACT coding mechanisms based on quantization, where less percep- tually important information is quantized more roughly and In this paper, we develop a multidimensional companding vice-versa, and lossless coding of the symbols emitted by the scheme for the preceptual distortion measure by S. van de Par quantizer to reduce statistical redundancy. In the quantization [1]. The scheme is asymptotically optimal in the sense that process of the (en)coder, it is desirable to quantize the infor- it has a vanishing rate-loss with increasing vector dimension. mation extracted from the signal in a rate-distortion optimal The compressor operates in the frequency domain: in its sim- sense, i.e., in a way that minimizes the perceptual distortion plest form, it pointwise multiplies the Discrete Fourier Trans- experienced by the user subject to the constraint of a certain form (DFT) of the windowed input signal by the square-root available bitrate. of the inverse of the masking threshold, and then goes back into the time domain with the inverse DFT. The expander is Due to its mathematical tractability, the Mean Square based on numerical methods: we do one iteration in a fixed- Error (MSE) is the elected choice for the distortion measure point equation, and then we fine-tune the result using Broy- in many coding applications. In audio coding, however, the den’s method. -
Model HR-ADC1 Analog to Digital Audio Converter
HALF-RACK SERIES Model HR-ADC1 Analog to Digital Audio Converter • Broadcast Quality Analog to Digital Audio Conversion • Peak Metering with Selectable Hold or Peak Store Modes • Inputs: Balanced and Unbalanced Stereo Audio • Operation Up to 24 bits, 192 kHz • Output: AES/EBU, Coaxial S/PDIF, AES-3ID • Selectable Internally Generated Sample Rate and Bit Depth • External Sync Inputs: AES/EBU, Coaxial S/PDIF, AES-3ID • Front-Panel External Sync Enable Selector and Indicator • Adjustable Audio Input Gain Trim • Sample Rate, Bit Depth, External Sync Lock Indicators • Peak or Average Ballistic Metering with Selectable 0 dB Reference • Transformer Isolated AES/EBU Output The HR-ADC1 is an RDL HALF-RACK product, featuring an all metal chassis and the advanced circuitry for which RDL products are known. HALF-RACKs may be operated free-standing using the included feet or may be conveniently rack mounted using available rack-mount adapters. APPLICATION: The HR-ADC1 is a broadcast quality A/D converter that provides exceptional audio accuracy and clarity with any audio source or style. Superior analog performance fine tuned through audiophile listening practices produces very low distortion (0.0006%), exceedingly low noise (-135 dB) and remarkably flat frequency response (+/- 0.05 dB). This performance is coupled with unparalleled metering flexibility and programmability with average mastering level monitoring and peak value storage. With external sync capability plus front-panel settings up to 24 bits and up to 192 kHz sample rate, the HR-ADC1 is the single instrument needed for A/D conversion in demanding applications. The HR-ADC1 accepts balanced +4 dBu or unbalanced -10 dBV stereo audio through rear-panel XLR or RCA jacks or through a detachable terminal block. -
Video Codec Requirements and Evaluation Methodology
Video Codec Requirements 47pt 30pt and Evaluation Methodology Color::white : LT Medium Font to be used by customers and : Arial www.huawei.com draft-filippov-netvc-requirements-01 Alexey Filippov, Huawei Technologies 35pt Contents Font to be used by customers and partners : • An overview of applications • Requirements 18pt • Evaluation methodology Font to be used by customers • Conclusions and partners : Slide 2 Page 2 35pt Applications Font to be used by customers and partners : • Internet Protocol Television (IPTV) • Video conferencing 18pt • Video sharing Font to be used by customers • Screencasting and partners : • Game streaming • Video monitoring / surveillance Slide 3 35pt Internet Protocol Television (IPTV) Font to be used by customers and partners : • Basic requirements: . Random access to pictures 18pt Random Access Period (RAP) should be kept small enough (approximately, 1-15 seconds); Font to be used by customers . Temporal (frame-rate) scalability; and partners : . Error robustness • Optional requirements: . resolution and quality (SNR) scalability Slide 4 35pt Internet Protocol Television (IPTV) Font to be used by customers and partners : Resolution Frame-rate, fps Picture access mode 2160p (4K),3840x2160 60 RA 18pt 1080p, 1920x1080 24, 50, 60 RA 1080i, 1920x1080 30 (60 fields per second) RA Font to be used by customers and partners : 720p, 1280x720 50, 60 RA 576p (EDTV), 720x576 25, 50 RA 576i (SDTV), 720x576 25, 30 RA 480p (EDTV), 720x480 50, 60 RA 480i (SDTV), 720x480 25, 30 RA Slide 5 35pt Video conferencing Font to be used by customers and partners : • Basic requirements: . Delay should be kept as low as possible 18pt The preferable and maximum delay values should be less than 100 ms and 350 ms, respectively Font to be used by customers . -
Audio Engineering Society Convention Paper
Audio Engineering Society Convention Paper Presented at the 128th Convention 2010 May 22–25 London, UK The papers at this Convention have been selected on the basis of a submitted abstract and extended precis that have been peer reviewed by at least two qualified anonymous reviewers. This convention paper has been reproduced from the author's advance manuscript, without editing, corrections, or consideration by the Review Board. The AES takes no responsibility for the contents. Additional papers may be obtained by sending request and remittance to Audio Engineering Society, 60 East 42nd Street, New York, New York 10165-2520, USA; also see www.aes.org. All rights reserved. Reproduction of this paper, or any portion thereof, is not permitted without direct permission from the Journal of the Audio Engineering Society. Loudness Normalization In The Age Of Portable Media Players Martin Wolters1, Harald Mundt1, and Jeffrey Riedmiller2 1 Dolby Germany GmbH, Nuremberg, Germany [email protected], [email protected] 2 Dolby Laboratories Inc., San Francisco, CA, USA [email protected] ABSTRACT In recent years, the increasing popularity of portable media devices among consumers has created new and unique audio challenges for content creators, distributors as well as device manufacturers. Many of the latest devices are capable of supporting a broad range of content types and media formats including those often associated with high quality (wider dynamic-range) experiences such as HDTV, Blu-ray or DVD. However, portable media devices are generally challenged in terms of maintaining consistent loudness and intelligibility across varying media and content types on either their internal speaker(s) and/or headphone outputs. -
PXC 550 Wireless Headphones
PXC 550 Wireless headphones Instruction Manual 2 | PXC 550 Contents Contents Important safety instructions ...................................................................................2 The PXC 550 Wireless headphones ...........................................................................4 Package includes ..........................................................................................................6 Product overview .........................................................................................................7 Overview of the headphones .................................................................................... 7 Overview of LED indicators ........................................................................................ 9 Overview of buttons and switches ........................................................................10 Overview of gesture controls ..................................................................................11 Overview of CapTune ................................................................................................12 Getting started ......................................................................................................... 14 Charging basics ..........................................................................................................14 Installing CapTune .....................................................................................................16 Pairing the headphones ...........................................................................................17 -
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/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. -
Image Compression Using Discrete Cosine Transform Method
Qusay Kanaan Kadhim, International Journal of Computer Science and Mobile Computing, Vol.5 Issue.9, September- 2016, pg. 186-192 Available Online at www.ijcsmc.com International Journal of Computer Science and Mobile Computing A Monthly Journal of Computer Science and Information Technology ISSN 2320–088X IMPACT FACTOR: 5.258 IJCSMC, Vol. 5, Issue. 9, September 2016, pg.186 – 192 Image Compression Using Discrete Cosine Transform Method Qusay Kanaan Kadhim Al-Yarmook University College / Computer Science Department, Iraq [email protected] ABSTRACT: The processing of digital images took a wide importance in the knowledge field in the last decades ago due to the rapid development in the communication techniques and the need to find and develop methods assist in enhancing and exploiting the image information. The field of digital images compression becomes an important field of digital images processing fields due to the need to exploit the available storage space as much as possible and reduce the time required to transmit the image. Baseline JPEG Standard technique is used in compression of images with 8-bit color depth. Basically, this scheme consists of seven operations which are the sampling, the partitioning, the transform, the quantization, the entropy coding and Huffman coding. First, the sampling process is used to reduce the size of the image and the number bits required to represent it. Next, the partitioning process is applied to the image to get (8×8) image block. Then, the discrete cosine transform is used to transform the image block data from spatial domain to frequency domain to make the data easy to process. -
MC14SM5567 PCM Codec-Filter
Product Preview Freescale Semiconductor, Inc. MC14SM5567/D Rev. 0, 4/2002 MC14SM5567 PCM Codec-Filter The MC14SM5567 is a per channel PCM Codec-Filter, designed to operate in both synchronous and asynchronous applications. This device 20 performs the voice digitization and reconstruction as well as the band 1 limiting and smoothing required for (A-Law) PCM systems. DW SUFFIX This device has an input operational amplifier whose output is the input SOG PACKAGE CASE 751D to the encoder section. The encoder section immediately low-pass filters the analog signal with an RC filter to eliminate very-high-frequency noise from being modulated down to the pass band by the switched capacitor filter. From the active R-C filter, the analog signal is converted to a differential signal. From this point, all analog signal processing is done differentially. This allows processing of an analog signal that is twice the amplitude allowed by a single-ended design, which reduces the significance of noise to both the inverted and non-inverted signal paths. Another advantage of this differential design is that noise injected via the power supplies is a common mode signal that is cancelled when the inverted and non-inverted signals are recombined. This dramatically improves the power supply rejection ratio. After the differential converter, a differential switched capacitor filter band passes the analog signal from 200 Hz to 3400 Hz before the signal is digitized by the differential compressing A/D converter. The decoder accepts PCM data and expands it using a differential D/A converter. The output of the D/A is low-pass filtered at 3400 Hz and sinX/X compensated by a differential switched capacitor filter. -
Rockbox User Manual
The Rockbox Manual for Sansa Fuze+ rockbox.org October 1, 2013 2 Rockbox http://www.rockbox.org/ Open Source Jukebox Firmware Rockbox and this manual is the collaborative effort of the Rockbox team and its contributors. See the appendix for a complete list of contributors. c 2003-2013 The Rockbox Team and its contributors, c 2004 Christi Alice Scarborough, c 2003 José Maria Garcia-Valdecasas Bernal & Peter Schlenker. Version unknown-131001. Built using pdfLATEX. Permission is granted to copy, distribute and/or modify this document under the terms of the GNU Free Documentation License, Version 1.2 or any later version published by the Free Software Foundation; with no Invariant Sec- tions, no Front-Cover Texts, and no Back-Cover Texts. A copy of the license is included in the section entitled “GNU Free Documentation License”. The Rockbox manual (version unknown-131001) Sansa Fuze+ Contents 3 Contents 1. Introduction 11 1.1. Welcome..................................... 11 1.2. Getting more help............................... 11 1.3. Naming conventions and marks........................ 12 2. Installation 13 2.1. Before Starting................................. 13 2.2. Installing Rockbox............................... 13 2.2.1. Automated Installation........................ 14 2.2.2. Manual Installation.......................... 15 2.2.3. Bootloader installation from Windows................ 16 2.2.4. Bootloader installation from Mac OS X and Linux......... 17 2.2.5. Finishing the install.......................... 17 2.2.6. Enabling Speech Support (optional)................. 17 2.3. Running Rockbox................................ 18 2.4. Updating Rockbox............................... 18 2.5. Uninstalling Rockbox............................. 18 2.5.1. Automatic Uninstallation....................... 18 2.5.2. Manual Uninstallation......................... 18 2.6. Troubleshooting................................. 18 3. Quick Start 20 3.1. -
A Multi-Frame PCA-Based Stereo Audio Coding Method
applied sciences Article A Multi-Frame PCA-Based Stereo Audio Coding Method Jing Wang *, Xiaohan Zhao, Xiang Xie and Jingming Kuang School of Information and Electronics, Beijing Institute of Technology, 100081 Beijing, China; [email protected] (X.Z.); [email protected] (X.X.); [email protected] (J.K.) * Correspondence: [email protected]; Tel.: +86-138-1015-0086 Received: 18 April 2018; Accepted: 9 June 2018; Published: 12 June 2018 Abstract: With the increasing demand for high quality audio, stereo audio coding has become more and more important. In this paper, a multi-frame coding method based on Principal Component Analysis (PCA) is proposed for the compression of audio signals, including both mono and stereo signals. The PCA-based method makes the input audio spectral coefficients into eigenvectors of covariance matrices and reduces coding bitrate by grouping such eigenvectors into fewer number of vectors. The multi-frame joint technique makes the PCA-based method more efficient and feasible. This paper also proposes a quantization method that utilizes Pyramid Vector Quantization (PVQ) to quantize the PCA matrices proposed in this paper with few bits. Parametric coding algorithms are also employed with PCA to ensure the high efficiency of the proposed audio codec. Subjective listening tests with Multiple Stimuli with Hidden Reference and Anchor (MUSHRA) have shown that the proposed PCA-based coding method is efficient at processing stereo audio. Keywords: stereo audio coding; Principal Component Analysis (PCA); multi-frame; Pyramid Vector Quantization (PVQ) 1. Introduction The goal of audio coding is to represent audio in digital form with as few bits as possible while maintaining the intelligibility and quality required for particular applications [1].