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MIGRATING RADIO CALL-IN TALK SHOWS to WIDEBAND AUDIO Radio Is the Original Social Network
Established 1961 2013 WBA Broadcasters Clinic Madison, WI MIGRATING RADIO CALL-IN TALK SHOWS TO WIDEBAND AUDIO Radio is the original Social Network • Serves local or national audience • Allows real-time commentary from the masses • The telephone becomes the medium • Telephone technical factors have limited the appeal of the radio “Social Network” Telephones have changed over the years But Telephone Sound has not changed (and has gotten worse) This is very bad for Radio Why do phones sound bad? • System designed for efficiency not comfort • Sampling rate of 8kHz chosen for all calls • 4 kHz max response • Enough for intelligibility • Loses depth, nuance, personality • Listener fatigue Why do phones sound so bad ? (cont) • Low end of telephone calls have intentional high- pass filtering • Meant to avoid AC power hum pickup in phone lines • Lose 2-1/2 Octaves of speech audio on low end • Not relevant for digital Why Phones Sound bad (cont) Los Angeles Times -- January 10, 2009 Verizon Communications Inc., the second-biggest U.S. telephone company, plans to do away with traditional phone lines within seven years as it moves to carry all calls over the Internet. An Internet-based service can be maintained at a fraction of the cost of a phone network and helps Verizon offer a greater range of services, Stratton said. "We've built our business over the years with circuit-switched voice being our bread and butter...but increasingly, we are in the business of selling, basically, data connectivity," Chief Marketing Officer John Stratton said. VoIP -
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. -
Polycom Voice
Level 1 Technical – Polycom Voice Level 1 Technical Polycom Voice Contents 1 - Glossary .......................................................................................................................... 2 2 - Polycom Voice Networks ................................................................................................. 3 Polycom UC Software ...................................................................................................... 3 Provisioning ..................................................................................................................... 3 3 - Key Features (Desktop and Conference Phones) ............................................................ 5 OpenSIP Integration ........................................................................................................ 5 Microsoft Integration ........................................................................................................ 5 Lync Qualification ............................................................................................................ 5 Better Together over Ethernet ......................................................................................... 5 BroadSoft UC-One Integration ......................................................................................... 5 Conference Link / Conference Link2 ................................................................................ 6 Polycom Desktop Connector .......................................................................................... -
NTT DOCOMO Technical Journal
3GPP EVS Codec for Unrivaled Speech Quality and Future Audio Communication over VoLTE EVS Speech Coding Standardization NTT DOCOMO has been engaged in the standardization of Research Laboratories Kimitaka Tsutsumi the 3GPP EVS codec, which is designed specifically for Kei Kikuiri VoLTE to further enhance speech quality, and has contributed to establishing a far-sighted strategy for making the EVS codec cover a variety of future communication services. Journal NTT DOCOMO has also proposed technical solutions that provide speech quality as high as FM radio broadcasts and that achieve both high coding efficiency and high audio quality not possible with any of the state-of-the-art speech codecs. The EVS codec will drive the emergence of a new style of speech communication entertainment that will combine BGM, sound effects, and voice in novel ways for mobile users. 2 Technical Band (AMR-WB)* [2] that is used in also encode music at high levels of quality 1. Introduction NTT DOCOMO’s VoLTE and that sup- and efficiency for non-real-time services, The launch of Voice over LTE (VoL- port wideband speech with a sampling 3GPP experts agreed to adopt high re- TE) services and flat-rate voice service frequency*3 of 16 kHz. In contrast, EVS quirements in the EVS codec for music has demonstrated the importance of high- has been designed to support super-wide- despite its main target of real-time com- quality telephony service to mobile users. band*4 speech with a sampling frequen- munication. Furthermore, considering In line with this trend, the 3rd Genera- cy of 32 kHz thereby achieving speech that telephony services using AMR-WB tion Partnership Project (3GPP) complet- of FM-radio quality*5. -
The Growing Importance of HD Voice in Applications the Growing Importance of HD Voice in Applications White Paper
White Paper The Growing Importance of HD Voice in Applications The Growing Importance of HD Voice in Applications White Paper Executive Summary A new excitement has entered the voice communications industry with the advent of wideband audio, commonly known as High Definition Voice (HD Voice). Although enterprises have gradually been moving to HD VoIP within their own networks, these networks have traditionally remained “islands” of HD because interoperability with other networks that also supported HD Voice has been difficult. With the introduction of HD Voice on mobile networks, which has been launched on numerous commercial mobile networks and many wireline VoIP networks worldwide, consumers can finally experience this new technology firsthand. Verizon, AT&T, T-Mobile, Deutsche Telekom, Orange and other mobile operators now offer HD Voice as a standard feature. Because mobile users tend to adopt new technology rapidly, replacing their mobile devices seemingly as fast as the newest models are released, and because landline VoIP speech is typically done via a headset, the growth of HD Voice continues to be high and in turn, the need for HD- capable applications will further accelerate. This white paper provides an introduction to HD Voice and discusses its current adoption rate and future potential, including use case examples which paint a picture that HD Voice upgrades to network infrastructure and applications will be seen as important, and perhaps as a necessity to many. 2 The Growing Importance of HD Voice in Applications White Paper Table of Contents What Is HD Voice? . 4 Where is HD Voice Being Deployed? . 4 Use Case Examples . -
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. -
HD Voice – a Revolution in Voice Communication
HD Voice – a revolution in voice communication Besides data capacity and coverage, which are one of the most important factors related to customers’ satisfaction in mobile telephony nowadays, we must not forget about the intrinsic characteristic of the mobile communication – the Voice. Ever since the nineties and the introduction of GSM there have not been much improvements in the area of voice communication and quality of sound has not seen any major changes. Smart Network going forward! Mobile phones made such a progress in recent years that they have almost replaced PCs, but their basic function, voice calls, is still irreplaceable and vital in mobile communication and it has to be seamless. In order to grow our customer satisfaction and expand our service portfolio, Smart Network engineers of Telenor Serbia have enabled HD Voice by introducing new network features and transitioning voice communication to all IP network. This transition delivers crystal-clear communication between the two parties greatly enhancing customer experience during voice communication over smartphones. Enough with the yelling into smartphones! HD Voice (or High-Definition Voice) represents a significant upgrade to sound quality in mobile communications. Thanks to this feature users experience clarity, smoothly reduced background noise and a feeling that the person they are talking to is standing right next to them or of "being in the same room" with the person on the other end of the phone line. On the more technical side, “HD Voice is essentially wideband audio technology, something that long has been used for conference calling and VoIP apps. Instead of limiting a call frequency to between 300 Hz and 3.4 kHz, a wideband audio call transmits at a range of 50 Hz to 7 kHz, or higher. -
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. -
Low Bit-Rate Speech Coding with Vq-Vae and a Wavenet Decoder
ICASSP 2019-2019 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), pp. 735-739. IEEE, 2019. DOI: 10.1109/ICASSP.2019.8683277. c 2019 IEEE. Personal use of this material is permitted. Permission from IEEE must be obtained for all other uses, in any current or future media, including reprinting/republishing this material for advertising or promotional purposes, creating new collective works, for resale or redistribution to servers or lists, or reuse of any copyrighted component of this work in other works. LOW BIT-RATE SPEECH CODING WITH VQ-VAE AND A WAVENET DECODER Cristina Garbaceaˆ 1,Aaron¨ van den Oord2, Yazhe Li2, Felicia S C Lim3, Alejandro Luebs3, Oriol Vinyals2, Thomas C Walters2 1University of Michigan, Ann Arbor, USA 2DeepMind, London, UK 3Google, San Francisco, USA ABSTRACT compute the true information rate of speech to be less than In order to efficiently transmit and store speech signals, 100 bps, yet current systems typically require a rate roughly speech codecs create a minimally redundant representation two orders of magnitude higher than this to produce good of the input signal which is then decoded at the receiver quality speech, suggesting that there is significant room for with the best possible perceptual quality. In this work we improvement in speech coding. demonstrate that a neural network architecture based on VQ- The WaveNet [8] text-to-speech model shows the power VAE with a WaveNet decoder can be used to perform very of learning from raw data to generate speech. Kleijn et al. [9] low bit-rate speech coding with high reconstruction qual- use a learned WaveNet decoder to produce audio comparable ity. -
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. -
EBU Evaluations of Multichannel Audio Codecs
EBU – TECH 3324 EBU Evaluations of Multichannel Audio Codecs Status: Report Source: D/MAE Geneva September 2007 1 Page intentionally left blank. This document is paginated for recto-verso printing Tech 3324 EBU evaluations of multichannel audio codecs Contents 1. Introduction ................................................................................................... 5 2. Participating Test Sites ..................................................................................... 6 3. Selected Codecs for Testing ............................................................................... 6 3.1 Phase 1 ....................................................................................................... 9 3.2 Phase 2 ...................................................................................................... 10 4. Codec Parameters...........................................................................................10 5. Test Sequences ..............................................................................................10 5.1 Phase 1 ...................................................................................................... 10 5.2 Phase 2 ...................................................................................................... 11 6. Encoding Process ............................................................................................12 6.1 Codecs....................................................................................................... 12 6.2 Verification of bit-rate -
Methods of Sound Data Compression \226 Comparison of Different Standards
See discussions, stats, and author profiles for this publication at: https://www.researchgate.net/publication/251996736 Methods of sound data compression — Comparison of different standards Article CITATIONS READS 2 151 2 authors, including: Wojciech Zabierowski Lodz University of Technology 123 PUBLICATIONS 96 CITATIONS SEE PROFILE Some of the authors of this publication are also working on these related projects: How to biuld correct web application View project All content following this page was uploaded by Wojciech Zabierowski on 11 June 2014. The user has requested enhancement of the downloaded file. 1 Methods of sound data compression – comparison of different standards Norbert Nowak, Wojciech Zabierowski Abstract - The following article is about the methods of multimedia devices, DVD movies, digital television, data sound data compression. The technological progress has transmission, the Internet, etc. facilitated the process of recording audio on different media such as CD-Audio. The development of audio Modeling and coding data compression has significantly made our lives One's requirements decide what type of compression he easier. In recent years, much has been achieved in the applies. However, the choice between lossy or lossless field of audio and speech compression. Many standards method also depends on other factors. One of the most have been established. They are characterized by more important is the characteristics of data that will be better sound quality at lower bitrate. It allows to record compressed. For instance, the same algorithm, which the same CD-Audio formats using "lossy" or lossless effectively compresses the text may be completely useless compression algorithms in order to reduce the amount in the case of video and sound compression.