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Perceptually-Motivated Objective Grading of Nonlinear Processing in Virtual-Bass Systems - November 2011
6 comments
The Loudness War: Do Louder, Hypercompressed Recordings Sell Better? - May 2011
12 comments
Improving the Magnitude Responses of Digital Filters for Loudspeaker Equalization - December 2010
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AES Engineering Briefs
Engineering Briefs are intended to be short verbal talks or poster presentations that will be of interest to AES members. Topics can be very wide-ranging, such as studio experience, equipment construction, new loudspeaker concepts, room acoustic measurements, analysis of audio equipment, and project studio startups, to name just a few. Relaxed reviewing of submissions will consider mainly whether they are of interest to AES members and are not overly commercial. The manuscripts will be restricted to 4 pages, and they will be available to all members for free download using the links on this page.
Authors: please read the Call for Engineering Briefs to see how you can submit an eBrief for AES Budapest this April.
Queen Mary's "Media and Arts Technology Studios" Audio System Design
The Media and Arts Technology Studios is a new facility at Queen Mary University of London linking together a previous Listening Room, with our new rooms: Control Room, Performance Lab, and Plant Room. This engineering report discusses our design philosophy for our given brief to create a "world-class facility" for a space that is a "blank canvas" for researchers. We detail considerations for making an audio system that is simple for standard recording and playback while having a tremendous amount of routing options for users to create unique projects between all the connected spaces, featuring two separate spatial audio reproduction systems. The result is a 96 kHz/24 bit MADI based system using a multimode fiber optic network and dedicated wordclock throughout.
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Revitalizing the Denis Arnold Hall for Multichannel Electroacoustic Sound Diffusion and Recording
The Denis Arnold Hall, the flagship lecturing and performance space in the Faculty of Music, Oxford University, has recently been the beneficiary of a complete refurbishment, including dedicated design and specification for the performance and diffusion of electroacoustic composition and multichannel sonic art. The new configuration includes acoustic absorption and insulation, low frequency management, and eight flexible full range satellite loudspeakers. This diffusion system is complemented by a full range of playback formats (from 1/4" reel-to-reel, to Blu-ray, with line level patching for discrete or "stemmed" multichannel performance/playback), as well as 16 small diaphragm capacitor microphones, hung in 8 stereo pairs from the ceiling on an integrated winch system. The design and configuration of the space necessitated a lengthy consultation with composers, acousticians, electricians, and audio-visual specialists, and lessons learned along the way might be useful for those interested in adapting their own space for similar purposes.
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Time Alignment of Subwoofers in Large PA Systems
In common PA systems the frequency range is divided into different ranges that are reproduced using different cabinets (subwoofers for the bass range and top cabinets for the mid-high range). This means different locations and positions of the sound sources and therefore notches and peaks in the crossover range. Time alignment is needed to adjust the arrival time of frequencies in the crossover area. But it's not only a matter of distance, since the crossover is modifying the phase, too. In this brief, a downscaled model is used to show how to measure the phase difference of a two-source PA system and correct it using FFT measuring software. Two situations are treated: two sources with overlapped frequency response and two sources sharing crossover frequency.
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Sound System Documentation for Construction Projects
This engineering brief will provide sound system engineers, users, and operators with the opportunity to become familiar with the information, processes, and documentation required to successfully coordinate the electrical, rough-in, and general construction requirements of a professionally installed sound system with the needs of architects, engineers, and contractors responsible for construction projects. The brief will include a categorized list of the 24 points of information that every architect, engineer, and construction professional needs to know about each and every sound system device to be implemented on a construction or renovation project. The presenter maintains 30 years of audio experience including front of house engineer for national recording artists, large scale integration as a sound systems contractor to complete professional design services for recital halls, performing arts centers, corporate, entertainment, and house of worship facilities.
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A Free Database of Head Related Impulse Response Measurements in the Horizontal Plane with Multiple Distances
A freely available database of Head-Related Impulse Response (HRIR) measurements is introduced. The impulse responses were measured in an anechoic chamber using a KEMAR manikin at four different loudspeaker distances—3 m, 2 m, 1 m, and 0.5 m—reaching from the far field to the near field. The loudspeaker was positioned at ear height and the manikin was rotated with a step motor in one degree increments. For the 3 m distance additional measurements have been carried out where the torso stayed fixed and only the artificial head was rotated. In addition to the raw impulse responses there are also data-sets available with a frequency response compensated for the use of several different headphones.
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Loudness Measurement and Human Interpretation in Television Program Quality Checking
Along the standardization of Loudness Measurement (EBU R128 and ITU-R BS.1770 update), France Television introduced loudness in 2010 to improve human interpretation during Quality Checking processes. After the defined measurement itself, done by a machine respecting the standard, studies have been conducted to fix objective limits according to the network's skills and viewer environments. By using the tools offered by the EBU R128, we have introduced specifications for audio acceptance beyond the loudness target of –23 LUFS, already taken into account. This work and results have been extended for common programs and short commercials in different ways. The measurement tools, developed according R128, have also been graphically developed to facilitate this additional reading.
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Do Young People Actually Care about the Quality of Their MP3s?
Ten focus groups were conducted in 2010 in Toronto, New York, and northern England, with participants ranging in age from 15 to 32 years old. Participants were asked about their file quality preferences for downloading music from online services, in the context of their existing positive/negative experiences, and being able to design a new service that would offer them the format(s) of their choice. This brief will outline some of the key qualitative findings, including: (1) Contrary to popular opinion, young people can tell the difference between a 128 k and 320 k MP3, but there appears to be an age threshold, below which listeners either do not notice or do not care. (2) Listening context and environment are important, and affect consumers’ file quality preferences, even when only considering what type of MP3 to download. (3) Consumers still find that they must weigh their personal quality preferences against other practical requirements.
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Creative Abuse in Time Stretching
An area of digital audio manipulation currently in flux is that of time stretching. Following the emergence of real-time granular synthesis as a compositional tool, early sampler-based implementations were pushed beyond "authenticity" to create new timbres in the commercial music of the 1990s. As the algorithms improve, allowing more flexible and transparent implementation today, even more opportunities for a new "creative abuse" exist. This brief will first contextualize through consideration of the metaphor of authenticity in the tape recording of the 1940s and its soon-parallel abuse, which offered new pathways into multi-tracking and Musique Concrète. The brief will chronologize, and continue by examining potential for exploitation of stretching artifacts in some contemporary algorithms, and propose a quantification of this effect.
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Activity Flow in Music Equalization: The Cognitive and Creative Implications of Interface Design
The mixing desk metaphor found in many Digital Audio Workstations (DAW) creates a quantitative visual display that is highly structured and segmented. While this is useful for transmitting large amounts of quantitative data it can inhibit the more intuitive and performative aspects inherent in music mixing. This paper's focus is the cognitive and creative issues encountered using current music production software equalizers and the influence they exert on the initial approach, task workflow, and final output of the user. Equalizers have been chosen to exemplify this, due to their pivotal balance between aural and visual modalities. The paper draws conclusions as to the effectiveness of current software equalizer designs and proposes modifications to design.
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The Preset Is Dead; Long Live the Preset
The use of preset sounds in audio production has long been scorned by professional producers, some of whom have cited a lack of originality or integrity, or perhaps a proliferation of homogenization in productions. Despite this, manufacturers have continued to develop ever-larger ranges of presets, now extending beyond instruments and effects, to EQ and even whole channel strips. Developmental work continues to further automate aspects of the mixing process itself. This brief will examine the implications of presets from yesterday to today, and using Logic Pro as a case study, offer some insight into the relevance of this evolving arena to the professional, and the implications to the enthusiast. It will conclude with some conjecture for the future.
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Automated Pure-Tone Audiometry Software Tool with Extended Frequency Range
In research institutes and other areas it is necessary to check the hearing of test subjects for listening tests (e.g., for assessing the quality of audio processing algorithms, etc.) by measuring pure tone audiograms. Often, neither a necessary skilled operator nor a special audiometer for high frequencies (>8kHz, important for the evaluation of a lot audio processing algorithms) are available. As an alternative, a software for automated audiogram measurements is presented here. This new kind of software runs on a standard PC with a high-quality sound card and audiological headphones and is operated by the test subject him/herself (self-screening). The implemented adaptive procedure allows fast standard audiogram measurements also for high frequencies up to 16 kHz. The challenges with respect to dynamic range and calibration are discussed.
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A Simple Reliable Power Amplifier with Minimal Component Count
We study an audio power amplifier that has three essential active components: two power MOSFETS and one operational amplifier. Such an amplifier will be reliable because MOSFETS have good safe-operating area properties, and there are no small semiconductors that require high voltage ratings. The topology is that of an op-amp directly driving a grounded-source complementary class-B MOSFET output stage. The center-tapped power supply for such an amplifier is floating, so each channel must have a separate supply, and there must be a small ±15-volt supply for the op-amp as well. We discuss the design and study the amplifier with simulations and an experimental prototype. It achieves good performance.
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Parametric Study of Magnet System Topologies for Microspeakers
A parametric simulation study was done in order to compare 3 different electrodynamic magnet system topologies: center-magnet, ring-magnet, and double-magnet (combined inner and outer magnet ring). The study is based on a row of simulations of the BL-factor (FEM and coil winding calculation), moving mass and effective radiating area of a microspeaker design where the inner coil diameter was changed. The dependency of the sound pressure level, the electrical quality factor, and the resonance frequency in a closed box on the inner coil diameter were derived to yield comparison charts for the 3 topologies.
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Signal Processing Applications for Automotive Audio
The automotive environment presents unique challenges to audio playback. Modern technology such as smart phones and high-efficiency power trains have created a new generation of audio sources and the need for audio systems capable of dealing with them. This Engineering Brief will introduce emerging automotive audio algorithms (listed below) and their design considerations.
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A New Golden Age of Recording
The “Golden Age” of recording is said to be in the early 1960s and consists of recordings that were made that have great natural sound that are a treat to the ear. I believe that the room is one of the major contributors to the sound of the “Golden Age” recordings. I have developed a technique for live recording producing good sound and also a means of adding quality room to multitrack recordings.
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What Constitutes Innovation in Music Production?
Innovation has often been at the core of record production, yet as production has advanced from Fred Gaisberg through the techniques of Musique Concrète to the plethora of possibilities afforded by the present digital age, the opportunities for genuine innovation might now seem limited. This notion is explored by considering the ontology of production with reference to audio examples, forming a chronological thread that highlights pieces commonly perceived as landmark innovations, their technological backdrops, and the recurrence/evolution of effect and aesthetic through successive generations of technology, and ultimately a nexus. The perception, attribution and value of “quality” is another factor, and while this is a separate subject in its own right, some discussion of this better contextualizes the topic.
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Achieving Great Sound in the Age of Loudness Wars
There has been much concern and discussion about the ever increasing demand for higher and higher mix and mastering levels. Veteran mastering engineers Adrian Carr and Bryan Martin will take a new perspective to discuss techniques and processes to improve and maintain fidelity in the current “Maximum-Volume-Level” marketplace. Though this presentation is geared toward the working engineers of today, it is of particular importance to the up and coming generation. Since the level of CD’s is not decreasing, the young engineer can still be aware of the advantages he has and the pitfalls he faces.
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Phantom Powering the Modern Condenser Microphone Part II: The Effects of Load Impedance on Microphone Performance
This paper builds on topics discussed in the previous AES paper “Phantom Powering the Modern Condenser Microphone: A Practical Look at Conditions for Optimized Performance.” It is not uncommon for microphone manufacturers to measure performance specifications using open load conditions. However, in application microphones are connected to mixing consoles that have much lower input impedances. These operating conditions can affect microphone performance and cause measurements to deviate from published open load conditions. During the previous investigation, changes were seen in several relevant microphone measurements as load impedance was changed. These observations prompted a more in-depth investigation into the effects of load impedance on microphone performance. The specific performance parameters investigated were: sensitivity, self-noise, current consumption, dynamic range, maximum sound pressure level (max SPL), signal to noise ratio (SNR) and frequency response.
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Digital Control of an Analog Parametric Equalizer
This project focuses on creating a digitally controllable analog band-pass filter with an adjustable resonant frequency for a middle frequency adjuster in an audio equalization stage. The design of the band-pass filter is a standard series resistor, inductor, and capacitor filter network. An adjustable gyrator circuit simulates an inductor to change the resonant frequency of the filter. Inside the gyrator circuit, a voltage-controlled amplifier is configured to simulate a resistor to change the gyrators simulated inductance. A digital to analog converter controls the gain of the voltage-controlled amplifier to make the analog filter digitally controlled. This circuit successfully acts as a bandpass filter with a digitally controllable resonant frequency.
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Mash-Up
The project will constitute an exploration of the choices made by users of audio post-production/editing applications in the selection and re-use of digital media files. The intended outcome would be to inform a better understanding of the ways in which contemporary users engage with digital media artifacts. The production context upon which this study will be based concerns an ongoing enterprise project, some further details of which are referred to below.
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Statistical Analysis of Electro-Acoustic Measurements Sets Using Scilab
The production management of electro-acoustic systems require the statistical analysis of measurements data. The analysis process should be sufficiently flexible to match the needs of the production process and the number of measured samples should be large enough to ensure the accuracy in statistical terms. Using an open source numerical computation software (Scilab) is possible to create statistical analysis procedures in a simple and cost effective way. Scilab syntax is simple enough to be acquired within a fairly short time, while data analysis capabilities are very advanced. In this work some sample applications are shown, with minimal code edit the provided examples can be adapted to several real world cases.
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Latency Measurements of Audio Sigma Delta Analog to Digital and Digital to Analog Converts
Latency is a well recognized issue when using digital audio workstations for live music processing. Previous research has reported measurements of the latency of the whole audio processing chain based on a “blackbox” approach. This report presents the results of latency measurement of typical compact analog to digital and digital to analog converters (ADC/DACs) in isolation from computer system processing overheads by using a high-speed data acquisition device. The report discusses the testing methods and pitfalls. It confirms that the latency is almost exclusively accounted for by the expected group delay of the digital decimation filters and interpolation filters used in the Sigma-Delta convertor.
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The Effect of Reverberation on Music Performance
The structure of playing musical instruments consists of 3 basic steps: performing on the instrument to make musical sounds, recognizing spatial information and music in the space through listening to the sounds themselves, and finally returning the information to the performance to adapt their musical experience and consciousness. These steps go on as a loop during the performance of musical instruments, and it is widely known that reverberation effects recognition of the space, i.e. the second step. Adjusting the whole acoustic environment inside the room with sampling reverberation should help musicians to play just as they want to. In this paper, the relation between multiple parameters of reverberation and the features of solo violin performance is investigated.
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Analysis-Synthesis Techniques for Additive Granular Synthesis
This project explores granular synthesis techniques that utilize various basis functions inspired by existing matching pursuit algorithms. The first algorithm performs a STFT on an input signal and synthesizes a new, granular signal using one-dimensional Gabor atoms. These atoms can be made to virtually reproduce the input signal, but a wide variety of granular effects can be achieved by altering the distribution of the atoms in the time and frequency domains, such as granular time stretching and pitch shifting, along with statistical distribution techniques introduced by the author. The second algorithm utilizes a basis set of generated noise bursts, which can be over-complete or an orthonormal basis for the Hilbert space that corresponds to the analysis window by applying the Gram-Schmidt process to the burst library. The noise functions are then used as grain contents in the synthesis stage, where a variety of effects are created with redistribution methods. Audio examples are provided over headphones.
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Harmonic Distortion Analysis in a Class-AB Tube Amplifier: The McIntosh MC-240
Some Class-AB tube amplifiers remain in demand for audiophiles due to their linear gain stages, low feedback, and minimal high-order harmonic distortion. The McIntosh MC240 Class-AB tube amplifier is a benchmark, high-quality stereo amplifier from the 1960s. This presentation gives an overview of the restoration of the amplifier to achieve a satisfactory level of performance, examination of the electrical topology in relation to the output signal’s distortion characteristics, and detailed analysis of the distortions produced by the amplifier using psychoacoustic-based listening tests as well as standard benchmark tests.
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The “Williams Star” Microphone Array Support System
With the ever increasing interest in 5-channel recording for home cinema, television, and pure 5-channel audio, the search for an operationally simple, reliable, and high quality surround sound recording system is becoming ever more important. The equal segment 4-channel and 5-channel arrays described in two AES papers by Michael Williams (AES preprints 3157 and 7480) are attracting more and more interest within the audio industry. However the research for a satisfactory operational microphone configuration cannot be dissociated from the purely mechanical problem of finding a suitable microphone array support system. The “Williams Star” Microphone Support System can provide a simple, flexible, and reliable microphone array support system for any equal segment array design. The visual impact of this system has also been reduced to an absolute minimum. Currently only the 4-channel and 5-channel seem to meet operational broadcasting requirements, but a 7-channel format is also provided for future developments with respect to the new Blu-ray format.
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Foreign-Language Dubbing Practices
The interdisciplinary nature of foreign language dubbing involves the collaboration of actors, directors, producers, translators, sound and video departments, and often develops into an elaborate production project in its own right. Increasingly, particularly with motion pictures, the dubbing industry faces new challenges as studios are required to work at the pace of the original production and follow any and all modifications, leading ultimately to the global release of a film everywhere simultaneously, an economic model that has been gathering momentum in recent years. The technology for foreign dialogue dubbing has undergone a great optimization of the production work-flow to accommodate such new requirements, and this presentation will show various practices currently in use throughout Europe and North America. I will discuss the idealized work-flow versus technology-specific solutions, cost optimization, media security, and the advantages of certain types of technologies on foreign-language synchronization. Areas of development in foreign-language dubbing and internationalization will be discussed.
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Application of Wave Field Synthesis and Analysis on Automotive Audio
High-quality audio equipment in cars enjoys rising popularity. For a remarkable number of people the car compartment represents the primary listening environment for enjoying music playback. Besides excellent sound quality spatial audio capabilities can also be expected from today’s top systems. In this talk we will give insight into the implementation of a spatial audio system based on Wave Field Synthesis that was realized inside an SUV. Audi AG and Fraunhofer IDMT together present the outcome of their research collaboration named "Audi Sound Concept." The Audi Q7 prototype has, on the one hand, remained a series production vehicle on the exterior, while on the other hand been rebuilt into a HiFi-Studio in the interior. The talk will include a systematic overview, a description of the playback-system, and some remarks about sound field analysis based on array measurements.
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Spooky Sounds: Interactive Audio Systems and Design for a Themed Attraction in an Academic Environment
The high-tech, interactive Gravesend Inn haunted hotel attraction features a large, distributed, audience-triggered sound system to implement a design both startling and evocative. Sound Designer Bruce Ellman and Systems Engineer John Huntington will discuss the challenges faced in developing this system and using it to teach Entertainment Technology Students.
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High Performance Architectural and Electro Acoustic Isolation Solutions
Increasing demands for community noise abatement, most specifically for architectural spaces such as night clubs and performance venue, have resulted in a variety of interesting design solutions to insure high performance sound attenuation, with FSTC results in excess of 80dB. This has often been accomplished with a combination of architectural construction design as well as electro-acoustic systems design such as directional subs, low frequency harmonic processing, and other systems integration devices. This paper will explore these designs and recent field results.
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A New Method for Evaluating Loudspeaker Efficiency in the Frequency Domain
The Constant Input Power (CIP) frequency response is proposed as a new method to evaluate loudspeaker efficiency in the frequency domain. Through a simulation study it is demonstrated how the CIP response can be a valuable tool when designing loudspeakers for which high efficiency is a priority.
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Wave Field Synthesis by Multiple Line Arrays
Wave field synthesis (WFS) is a spatial audio rendering technique that produces a physical approximation of wavefronts for virtual sources. Large loudspeaker arrays can simulate a virtual source that exists outside of the listening room. The technique is traditionally limited to the horizontal plane due to the prohibitive cost of planar loudspeaker arrays. Multiple-line-array wave field synthesis is proposed as an extension to linear WFS. This method extends the virtual source space in the vertical direction using a fraction of the number of loudspeakers required for plane arrays. This paper describes a listening test and software environment capable of driving a loudspeaker array according to the proposed extension, as well as the construction of a modular loudspeaker array that can be adapted to multiple-line configurations.
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Playback Disappointment in Linear PCM Recording Systems
As an “in-the-trenches” music recording engineer, my workflow has evolved to essentially an all in-the-box approach, with the exception of some external DSP. After spending many hours on a mix, I always get a strong feeling that when I finally print my mix to a stereo track in my workstation, or to a hardware-based digital recorder and play back the resulting 24-Bit-96-kHz WAV file, that it’s just not the same. A sense of disappointment. It seems to lack depth, reverb tails fall off, the transient response seems dulled, and an overall “graininess” to the mix. This paper and presentation will demonstrate the differences by way of 5.6-MHz DSD null tests and explore the difference between a live digital stream, and the disappointing playback of that digital stream that has been captured by a recorder in WAV file format. Upon demonstrating the problem, I will discuss possible solutions and workarounds I have used.
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Consumer Attitudes Toward Digital Audio Quality
This paper builds upon an engineering brief submitted to the 130th AES Convention (Harris 2011). Where the May 2011 brief outlined initial findings from focus groups that were conducted, considering questions about preferred audio quality from the point of view of attitudes and consumer behavior, this brief focuses on an outline for future research, discussing important questions for consideration, and proposed methodology.
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The Effect of Downmixing on Measured Loudness
ITU-R BS.1770 has become the standard for loudness measurement in broadcasting. The measurement algorithm is equally adapted to 5.1 channel audio signals as it to a 2-channel downmix. Due to the manner by which the channels are summed, loudness differences can occur between the 5.1 channel signal and that of the stereo downmix. These differences are dependent on the inter-channel characteristics of the 5-channel mix. This engineering brief will outline the differences that can occur with different signals and provide data using real-world broadcast signals.
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Prediction of Valence and Arousal from Music Features
Mood is an important attribute of music, and knowledge on mood can be used as a basic ingredient in music recommender and retrieval systems. Moods are assumed to be dominantly determined by two dimensions: valence and arousal. An experiment was conducted to attain data for song-based ratings of valence and arousal. It is shown that subject-averaged valence and arousal can be predicted from music features by a linear model.
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The Engineering Briefs at this Convention have been selected on the basis of a submitted synopsis, ensuring that they are of interest to AES members, and are not overly commercial. These briefs have been reproduced from the authors' advance manuscripts, without editing, corrections, or consideration by the Review Board. The AES takes no responsibility for their contents. Paper copies are not available, but any member can freely access these briefs. Members are encouraged to provide comments that enhance their usefulness.
