Recording and Production

Thursday, October 19, 2:00 pm — 6:00 pm (Rm 1E11)

Chair:
Stephen Roessner, University of Rochester - Rochester, NY, USA

P09-1 Analysis and Prediction of the Audio Feature Space when Mixing Raw Recordings into Individual Stems—Marco A. Martinez Ramirez, Queen Mary University of London - London, UK; Joshua D. Reiss, Queen Mary University of London - London, UK
Processing individual stems from raw recordings is one of the first steps of multitrack audio mixing. In this work we explore which set of low-level audio features are sufficient to design a prediction model for this transformation. We extract a large set of audio features from bass, guitar, vocal, and keys raw recordings and stems. We show that a procedure based on random forests classifiers can lead us to reduce significantly the number of features and we use the selected audio features to train various multi-output regression models. Thus, we investigate stem processing as a content-based transformation, where the inherent content of raw recordings leads us to predict the change of feature values that occurred within the transformation. 

P09-2 The Beat Goes Static: A Tempo Analysis of U.S. Billboard Hot 100 #1 Songs from 1955–2015—Stephen Roessner, University of Rochester - Rochester, NY, USA
The Billboard Hot 100 is a rich source of information for tracking musical trends. Using available data analysis tools, we devised a method to accurately track tempo throughout a song. In this paper we demonstrate through an analysis of all number one songs from the chart that tempo variation within a song has declined over a 60-year period. In the 5-year span from 1955–1959, the average standard deviation of tempo was 5.01 beats per minute, or about 4.8%. Conversely, from 2010–2014, the average standard deviation was less than 1 beat per minute, or only about 0.85% of the average tempo. 

P09-3 An Even-Order Harmonics Control Technique for Analog Pedal Effector—Kanako Takemoto, Hiroshima Institute of Technology - Hiroshima, Japan; Shiori Oshimo, Hiroshima Institute of Technology - Hiroshima, Hiroshima-ken, Japan; Toshihiko Hamasaki, Hiroshima Institute of Technology - Hiroshima, Japan
The primary distortion mechanism of the analog guitar pedal effector is saturating nonlinearity of a transfer function, which consists of operational amplifier and diode clippers with filters. The output spectrum of this system shows odd-order harmonics primarily, but it also contains even-order harmonics. We found that the intensity of this even-order harmonic varies depending on the power supply voltage and clarified the mechanism by analysis of the internal circuit topology of the operational amplifier. The analysis was justified compared with the conventional single-ended transistor pedal operation. Based on the analysis we proposed a new concept of even harmonic control technique, which was applied for analog “Distortion” pedals and demonstrated distinguished experimental results with a prototype. 

P09-4 Unified Modeling for Series of Miniature Twin Triode Tube—Shiori Oshimo, Hiroshima Institute of Technology - Hiroshima, Hiroshima-ken, Japan; Kanako Takemoto, Hiroshima Institute of Technology - Hiroshima, Japan; Toshihiko Hamasaki, Hiroshima Institute of Technology - Hiroshima, Japan
Unification of high precision SPICE modeling for the series of MT vacuum tube has been succeeful for the first time. The model formula was validated based on the comparison of electrode and space physical dimensions among 12AX7, 12AU7, 12AY7, and 12AT7 associated with various aspects of properties not limited to the general Ip-Vp family curves. As a result, the non-linear behavior of the grid current and the plate current as a function of plus/minus grid voltage were able to be expressed entirely by 17 parameters of the newly proposed SPICE model, in which 4 tube type specific parameters  and 4 universal parameters are constant and matching of twin valves of each tube as well as product dispersion are fitted by 9 variable parameters. 

P09-5 Virtual Analog Modeling of a UREI 1176LN Dynamic Range Control System—Etienne Gerat, Helmut Schmidt University Hamburg - Hamburg, Germany; Felix Eichas, Helmut Schmidt University - Hamburg - Hamburg, Germany; Udo Zölzer, Helmut-Schmidt-University Hamburg - Hamburg, Germany
This paper discusses an application of block-oriented modeling to a popular analog dynamic range compressor using iterative minimization. The reference device studied here is the UREI 1176LN, which has been widely used in music production and recording. A clone of the circuit built in a previous project has been used as a reference device to compare the results of the implementation. A parametric block-oriented model has been designed, improved, and tuned using the Levenberg-Marquardt iterative error minimization algorithm. Only input/output measurements have been performed following a gray-box modeling approach. Finally the model has been evaluated with objective scores and a listening test. This work led to very convincing modeling results. 

P09-6 Amplitude Panning and the Interior Pan—Mark R. Thomas, Dolby Laboratories - San Francisco, CA, USA; Charles Q. Robinson, Dolby Laboratories - San Francisco, CA, USA
The perception of source location using multi-loudspeaker amplitude panning is considered. While there exist many perceptual models for pairwise panning, relatively few studies consider the general multi-loudspeaker case. This paper evaluates panning scenarios in which a source is panned on the boundary or within the volume bounded by discrete loudspeakers, referred to as boundary and interior pans respectively. Listening results reveal the following: (1) pans to a single loudspeaker yield lowest localization error, (2) pairwise pans tend to be consistently localized closer to the listener than single loudspeaker pans, (3) largest errors occur when the virtual source is panned close to the listener, (4) interior pans are accurately perceived and, surprisingly, in some cases more accurately than pairwise pans. 

P09-7 Recording in a Virtual Acoustic Environment—Jonathan S. Abel, Stanford University - Stanford, CA, USA; Elliot K. Canfield-Dafilou, Center for Computer Research in Music and Acosutics (CCRMA), Stanford University - Stanford, CA, USA
A method is presented for high-quality recording of voice and acoustic instruments in loudspeaker-generated virtual acoustics. Auralization systems typically employ close-mic'ing to avoid feedback, while classical recording methods prefer high-quality room microphones to capture the instruments integrated with the space. Popular music production records dry tracks, and applies reverberation after primary edits are complete. Here a hybrid approach is taken, using close mics to produce real-time, loudspeaker-projected virtual acoustics, and room microphones to capture a balanced, natural sound. The known loudspeaker signals are then used to cancel the virtual acoustics from the room microphone tracks, providing a set of relatively dry tracks for use in editing and post-production. Example recordings of Byzantine chant in a virtual Hagia Sophia are described. 

P09-8 A Study of Listener Bass and Loudness Preferences over Loudspeakers and Headphones—Elisabeth McMullin, Samsung Research America - Valencia, CA USA
In order to study listener bass and loudness preferences over loudspeakers and headphones a series experiments using a method of adjustment were run. Listeners adjusted the bass and loudness levels of multiple genres of music to their personal preference in separate listening sessions over loudspeakers in a listening room and headphones equalized to simulate loudspeakers in a listening room. The results indicated that listeners who preferred more bass over both headphones and loudspeakers also tended to listen at higher levels. Furthermore the majority of listeners preferred slightly higher bass and loudness levels over loudspeakers than over headphones. Listener factors including musical preferences, hearing ability, and training level are also explored.