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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. -
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. -
Transition to Digital Digital Handbook
TRANSITION TO DIGITAL DIGITAL HANDBOOK Audio's video issues tion within one audio sample, the channels of audio over a fiber-optic in- There can be advantages to locking preambles present a unique sequence terface. This has since been superseded the audio and video clocks, such as for (which violate the Biphase Markby AES 10 (or MADI, Multichannel editing, especially when the audio and Code) but nonetheless are DC -freeAudio Digital Interface), which sup- video programs are related. Althoughand provide clock recovery. ports serial digital transmission of 28, digital audio equipment may provide 56, or 64 channels over coaxial cable or an analog video input, it is usually bet- Like AES3, but not fiber-optic lines, with sampling rates ter to synchronize both the audio and A consumer version of AES3 -of up to 96kHz and resolution of up to the video to a single higher -frequencycalled S/PDIF, for Sony/Philips Digi-24 bits per channel. The link to the IT source, such as a 10MHz master refer-tal Interface Format (more formallyworld has also been established with ence. This is because the former solu- known as IEC 958 type II, part of IEC- AES47, which specifies a method for tion requires a synchronization circuit60958) - is also widely used. Essen- packing AES3 streams over Asynchro- that will introduce some jitter into thetially identical to AES3 at the protocol nous Transfer Mode (ATM) networks. signal, especially because the video it- level, the interface uses consumer - It's also worth mentioning Musical self may already have some jitter. To ac- friendly RCA jacks and coaxial cable.Instrument Digital Interface (MIDI) for broadcast operations. -
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. -
Asi6614, Asi6618 Multistream Pci-Express Sound Card
09 DECEMBER 07 ASI6614, ASI6618 MULTISTREAM PCI-EXPRESS SOUND CARD DESCRIPTION FEATURES The ASI6614 and ASI6618 are professional PCI-Express sound cards • 4 or 12 mono/stereo streams of playback into 4 stereo outputs designed for use in radio broadcast automation. (ASI6614) Providing up to 16 play streams that are mixed to 4 (ASI6614) or 8 • 8 or 16 mono/stereo streams of playback into 8 stereo outputs (ASI6618) stereo outputs and up to 2 record streams fed from one (ASI6618) stereo input, the ASI6614 and ASI6518 feature AudioScience’s unique “anything to anywhere” mixing and routing. • 1 or 2 mono/stereo streams of record from 1 stereo input The ASI6614 and ASI6618 provide both balanced analog and AES/EBU • Formats include PCM, MPEG layer 2 and MP3 with sample rates to inputs and outputs. The maximum analog input and output level is 96kHz +24dBu. • MRX™ technology supports digital mixing of multiple stream A choice of uncompressed PCM, MPEG layer 2 and MP3 is available for formats and sample rates both recording and playback. All compression is handled by an on- board floating point DSP, allowing the host computer to focus on other • TSX™ time scaling allows compression/expansion of play streams tasks. by up to +/-20% with no pitch shift ASI6614 and ASI6618 functionality includes MRX™ multi-rate mixing • SSX™ mode for multichannel playback and mixing technology that allows streams of different sample-rates and formats to be mixed digitally. TSX™ time scaling allows compression/expansion of • Balanced stereo analog inputs and outputs with levels to +24dBu any or all playback streams in real time with no change in pitch. -
Le Décibel ( Db ) Est Un Sous-Multiple Du Bel, Correspondant À 1 Dixième De Bel
Emploi du Décibel, unité relative ou unité absolue Le décibel ( dB ) est un sous-multiple du bel, correspondant à 1 dixième de bel. Nommé en l’honneur de l'inventeur Alexandre Graham Bell, le bel est unité de mesure logarithmique du rapport entre deux puissances, connue pour exprimer la puissance du son. Grandeur sans dimension en dehors du système international, le bel n'est pas l'unité la plus fréquente. Le décibel est plus couramment employé. Équivalent à 1/10 de bel, le décibel comme le bel, peut être utilisé dans les domaines de l’acoustique, de la physique, de l’électronique et est largement répandue dans l’ensemble des champs de l’ingénierie (fiabilité, inférence bayésienne, etc.). Cette unité est particulièrement pertinente dans les domaines où la perception humaine est mise en jeu, car la loi de Weber-Fechner stipule que la sensation ressentie varie comme le logarithme de l’excitation. où S est la sensation perçue, I l'intensité de la stimulation et k une constante. Histoire des bels et décibels Le bel (symbole B) est utilisé dans les télécommunications, l’électronique, l’acoustique ainsi que les mathématiques. Inventé par des ingénieurs des Laboratoires Bell pour mesurer l’atténuation du signal audio sur une distance d’un mile (1,6 km), longueur standard d’un câble de téléphone, il était appelé unité de « transmission » à l’origine, ou TU (Transmission unit ), mais fut renommé en 1923 ou 1924 en l’honneur du fondateur du laboratoire et pionnier des télécoms, Alexander Graham Bell . Définition, usage en unité relative Si on appelle X le rapport de deux puissances P1 et P0, la valeur de X en bel (B) s’écrit : On peut également exprimer X dans un sous multiple du bel, le décibel (dB) un décibel étant égal à un dixième de bel.: Si le rapport entre les deux puissances est de : 102 = 100, cela correspond à 2 bels ou 20 dB. -
Application Note
AN5073 Application note Receiving S/PDIF audio stream with the STM32F4/F7/H7 Series Introduction The Sony/Philips Digital Interface Format (S/PDIF) is a point-to-point protocol for serial and uni-directional transmission of digital audio through a single transmission line for consumer and professional applications. The transmission of data can be done in several ways, by electrical or optical means. The S/PDIFRX peripheral embedded in STM32 devices is designed to receive an S/PDIF flow compliant with IEC-60958 and IEC-61937, which define the physical implementation requirements as well as the coding and the protocol. These standards support simple stereo streams up to high sample rates, and compressed multi-channel surround sound, such as those defined by Dolby or DTS. This application note describes electrical interfaces, to properly connect the S/PDIF stream generated by an external device to an STM32 device embedding the S/PDIFRX interface peripheral, since the voltage level of the S/PDIF line is not the same as that used in STM32 devices. AN5073 - Rev 2.0 - June 2018 www.st.com For further information contact your local STMicroelectronics sales office. AN5073 S/PDIF Interface 1 S/PDIF Interface This document applies to Arm®-based devices. Note: Arm is a registered trademark of Arm Limited (or its subsidiaries) in the US and/or elsewhere. 1.1 S/PDIF background S/PDIF is an audio interface for transmission of digital audio data over reasonably short distances between modules of systems such as home theaters or hi-fi. S/PDIF is a single-wire serial uni-directional, self-clocking interface. -
Sd/Hd Mpeg 2 and Mpeg 4 Ird
MODEL RD-RD-3030 SD/HD MPEG 2 AND MPEG 4 IRD www.amt.com RD-30 SD/HD MPEG 2 and MPEG 4 IRD Video Decoder Profiles: * VBI / VANC PROCESSING MPEG 2 SD/HD Profile : ISO 13818-2 MP@ML and MP@HL SDI Ancillary support for: MPEG 4 SD/HD Profile: ISO/IEC 14496-10 MP@L3 [email protected] Closed Captioning (CEA-708), AFD (SMPTE 2016), Supported Resolutions: OP-47 (SMPTE RDD-08), SMPTE RDD-11, 480i59.94, 576i50, 720p50/59.94/60, SCTE 127 (SMPTE 2031), EN301775 (SMPTE 2031), 1080i50/59.94/60, 1080p23.97/24/25/29.97/30 Time Code (SMPTE 12M-2), SCTE 35 to SCTE 104 Conversion Inputs CVBS/SDI VBI waveform support: Sync Input: Auto-detects Bi-Level and Tri-level sync for Genlock Connector: 1x BNC (75 Ohm) Line 21 captions (CEA-608), TVG2x, AMOL-48/96 (SCTE 127), DVB-ASI Teletext/WSS/VPS (EN 301775) Interface: ASI (EN 50083-9 250 kb/s to 200 Mb/s) Connector: 1x BNC (75 Ohm) IP DVB-S/S2 Hardware Option (RD30-01-LB) - Optional* Standard: UDP, RTP and SMPTE 2022-1 2007 FEC DVB-CI per EN 50221 Supports 1 to 7 TS packets per IP packet Interface: 2x DVB-CI CAM slots IGMP v1, v2 and v3 support Use: De-scrambles decoded service only without Input TS Rate: 250 kb/s to 150 Mb/s purchase of RD30-CAM-KEY. RD30-CAM-KEY Connection speed: GigE (10/100/1000 Auto-Negotiate) Receiver capability: 2 simultaneous MPEG over IP transport streams provides up to maximum amount of decryptable Connector: 2x RJ45 services supported by CAM module. -
A 449 MHZ MODULAR WIND PROFILER RADAR SYSTEM by BRADLEY JAMES LINDSETH B.S., Washington University in St
A 449 MHZ MODULAR WIND PROFILER RADAR SYSTEM by BRADLEY JAMES LINDSETH B.S., Washington University in St. Louis, 2002 M.S., Washington University in St. Louis, 2005 A thesis submitted to the Faculty of the Graduate School of the University of Colorado in partial fulfillment of the requirement for the degree of Doctor of Philosophy Department of Electrical, Computer, and Energy Engineering 2012 This thesis entitled: A 449 MHz Modular Wind Profiler Radar System written by Bradley James Lindseth has been approved for the Department of Electrical, Computer, and Energy Engineering Prof. Zoya Popović Dr. William O.J. Brown Date The final copy of this thesis has been examined by the signatories, and we Find that both the content and the form meet acceptable presentation standards Of scholarly work in the above mentioned discipline. Lindseth, Bradley James (Ph.D., Electrical Engineering) A 449 MHz Modular Wind Profiler Radar System Thesis directed by Professor Zoya Popović and Dr. William O.J. Brown This thesis presents the design of a 449 MHz radar for wind profiling, with a focus on modularity, antenna sidelobe reduction, and solid-state transmitter design. It is one of the first wind profiler radars to use low-cost LDMOS power amplifiers combined with spaced antennas. The system is portable and designed for 2-3 month deployments. The transmitter power amplifier consists of multiple 1-kW peak power modules which feed 54 antenna elements arranged in a hexagonal array, scalable directly to 126 elements. The power amplifier is operated in pulsed mode with a 10% duty cycle at 63% drain efficiency. -
Sox Examples
Signal Analysis Young Won Lim 2/17/18 Copyright (c) 2016 – 2018 Young W. Lim. 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 Sections, no Front-Cover Texts, and no Back-Cover Texts. A copy of the license is included in the section entitled "GNU Free Documentation License". Please send corrections (or suggestions) to [email protected]. This document was produced by using LibreOffice. Young Won Lim 2/17/18 Based on Signal Processing with Free Software : Practical Experiments F. Auger Audio Signal Young Won Lim Analysis (1A) 3 2/17/18 Sox Examples Audio Signal Young Won Lim Analysis (1A) 4 2/17/18 soxi soxi s1.mp3 soxi s1.mp3 > s1_info.txt Input File Channels Sample Rate Precision Duration File Siz Bit Rate Sample Encoding Audio Signal Young Won Lim Analysis (1A) 5 2/17/18 Generating signals using sox sox -n s1.mp3 synth 3.5 sine 440 sox -n s2.wav synth 90000s sine 660:1000 sox -n s3.mp3 synth 1:20 triangle 440 sox -n s4.mp3 synth 1:20 trapezium 440 sox -V4 -n s5.mp3 synth 6 square 440 0 0 40 sox -n s6.mp3 synth 5 noise Audio Signal Young Won Lim Analysis (1A) 6 2/17/18 stat Sox s1.mp3 -n stat Sox s1.mp3 -n stat > s1_info_stat.txt Samples read Length (seconds) Scaled by Maximum amplitude Minimum amplitude Midline amplitude Mean norm Mean amplitude RMS amplitude Maximum delta Minimum delta Mean delta RMS delta Rough frequency Volume adjustment Audio Signal Young Won -
MT-230 One Technology Way • P
Mini Tutorial MT-230 One Technology Way • P. O. Box 9106 • Norwood, MA 02062-9106, U.S.A. • Tel: 781.329.4700 • Fax: 781.461.3113 • www.analog.com Because this headroom of the converter is continuously being Noise Considerations in High constrained, maintaining a very low noise spectral density of Speed Converter Signal Chains −150 dBFS/Hz and lower is challenging with each new design. This is why it is paramount that the designer recognize the by Rob Reeder Analog Devices, Inc. importance of the surrounding noise contributions within the entire signal chain solution. Admittedly, there are many noise principles. This tutorial IN THIS MINI TUTORIAL addresses two of these principles: noise bandwidth and the The most common outside noise sources and how they addition of noise sources. influence the total dynamic system performance of a high NOISE BANDWIDTH speed signal chain are described as well as some analog and Noise bandwidth is not the same as the typical −3 dB band- digital tricks that can be employed to further increase the width of an amplifier or filter cutoff point. The shape for noise signal-to-noise ratio (SNR) in your next design. takes on a different form (rectangular) which specifies the total integration of the bandwidth. This means making a slight INTRODUCTION adjustment in the noise calculation when considering the noise Designing in high speed analog signal chains can be challenging bandwidth contribution. with so many noise sources to consider. Whether the frequen- For a first-order system, for example a first-order low-pass cies are high speed (>10 MHz) or low speed, the converter filter, the noise bandwidth is 57% larger. -
Audio Engineering Society Standards Committee
Audio Engineering Society Standards Committee Notice and DRAFT agenda for the meeting of the SC-02-02 Working Group on digital input/output interfacing of the SC-02 Subcommittee on Digital Audio To be held in conjunction with the upcoming AES 149th Convention. The meeting is scheduled to take place online, 2020-10. Please check the latest schedule at: http://www.aes.org/standards/ 1. Formal notice on patent policy 2. Introduction to working group and attendees 3. Amendments to and approval of agenda Note that projects where there is no current proposal for revision or amendment, and where there is at least 12 months before any formal review is due, are listed in an annex to this agenda. Please let the chair know if you propose to discuss any projects in this annex. 4. Approval of report of previous meeting, held online, 2020-05. 5. Open Projects NOTE: One or more of these projects may be in the process of a formal Call for Comment (CFC), as indicated by the project status. In these cases only, due process requires that any comments be published. AES10-R Review of AES10-2008 (r2019): AES Recommended Practice for Digital Audio SC-02-02 Engineering - Serial Multichannel Audio Digital Interface (MADI) scope: This standard describes the data organization and electrical characteristics for a multichannel audio digital interface (MADI). It includes a bit-level description, features in common with the two-channel format of the AES3, AES Recommended Practice for Digital Audio Engineering - Serial Transmission Format for Linearly Represented Digital Audio Data, and the data rates required for its utilization.