AES New York 2018
Poster Session P03
P03 - Recording and Production
Wednesday, October 17, 10:00 am — 11:30 am
P03-1 Characterizing the Effect on Linear and Harmonic Distortions of AC Bias and Input Levels when Recording to Analog Tape—Thomas Mitchell, University of Miami - Coral Gables, FL, USA; Christopher Bennett, University of Miami - Coral Gables, FL, USA
Analog tape recorders introduce both linear distortions and nonlinear distortions to the audio. While the role of the AC bias and input levels on these distortions are well-understood by recording engineers, the impact on specific audio features, for example SNR, fatness, brightness, roughness, and harmonic count is less well described. In this study we examined with high granularity the impact and interactions of several AC bias and input levels on each of these features. We utilized the exponential swept sine acquisition and deconvolution technique to analyze a Scully 280. The results provide a detailed characterization on the tonal character introduced by the recorder. We conclude with level recommendations that could prove important for primary capture, effects processing, and digital emulation.
Convention Paper 10039
P03-2 Microphone Comparison for Snare Drum Recording—Matthew Cheshire, Birmingham City University - Birmingham, UK; Jason Hockman, Birmingham City University - Birmingham, UK; Ryan Stables, Birmingham City University - Birmingham, UK
We present two experiments to test listener preference for snare microphones within real-world recording scenarios. In the first experiment, listeners evaluated isolated recordings captured with 25 microphones. In the second experiment, listeners performed the same task with the addition of a kick drum and hi-hat as part of a performed drum sequence. Results indicate a prominent contrast between the highest and lowest rated microphones and that condensers were rated higher than other subsets tested. The preference for three microphones significantly changed between the two listening test conditions. A post-test survey revealed that most listeners compared high-frequency characteristics, which were measured using spectral features. A positive correlation was observed between test scores of cardioid microphones and the brightness feature.
Convention Paper 10040
P03-3 Automatic Mixing of Multitrack Material Using Modified Loudness Models—Steven Fenton, University of Huddersfield - Huddersfield, West Yorkshire, UK
This work investigates the perceptual accuracy of the ITU-Recommendation BS.1770 loudness algorithm when employed in a basic auto mixing system. Optimal filter parameters previously proposed by the author, which incorporate modifications to both the pre-filter response and the integration window sizes are tested against the standard K-weighted model and filter parameters proposed through other studies. The validation process encompassed two stages, the first being the elicitation of preferred mix parameters used by the mixing system and the subsequent generation of automatic mixes based on these rules utilizing the various filter parameters. A controlled listening test was then employed to evaluate the listener preferences to the completed mixes. It is concluded that the optimized filter parameter set based upon stem type, results in a more perceptually accurate automatic mix being achieved.
Convention Paper 10041
P03-4 Double-MS Decoding with Diffuse Sound Control—Alexis Favrot, Illusonic GmbH - Uster, Switzerland; Christof Faller, Illusonic GmbH - Uster, Zürich, Switzerland; EPFL - Lausanne, Switzerland; Helmut Wittek, SCHOEPS GmbH - Karlsruhe, Germany
The double MS (DMS) setup provides a coincident recording configuration in the horizontal plane for surround sound recording, similar to Ambisonics B-format. DMS uses two cardioids and one dipole microphones, arranged coincidentally. An algorithm for processing DMS recordings is described, which in addition to linear processing provides a diffuse sound gain control and diffuse sound de-correlation. The target stereo or multichannel signals can be made more dry or reverberant with diffuse gain. Diffuse de-correlation improves spaciousness and its importance scales with the number of channels. The consequences of these new controls for the stereophonic image will be depicted.
Convention Paper 10042
P03-5 Real-Time System for the Measurement of Perceived Punch—Andrew Parker, University of Huddersfield - Huddersfield, UK; Steven Fenton, University of Huddersfield - Huddersfield, West Yorkshire, UK; Hyunkook Lee, University of Huddersfield - Huddersfield, UK
Previous work has proposed a perceptually motivated model for the objective measurement of punch in a music recording. This paper presents a real-time implementation of the proposed model as a punch metering plug-in. The plug-in presents both momentary and historical punch metrics in the form of a P95 (95th percentile punch measure), Mean punch score, and P95M (95th percentile punch divided by the mean punch score). The meter’s outputs are compared to subjective punch scores derived through a controlled listening test and show a “strong” correlation with Pearson and Spearman coefficients r=0.840 (p<0.001) and rho=0.937 (p<0.001) respectively. The real-time measure of punch could prove useful in allowing objective control and optimization of this feature during mixing, mastering and broadcast.
Convention Paper 10043
P03-6 Algorithm to Determine Downmixing Coefficients from Specific Multichannel Format to Reproduction Format with a Smaller Number of Channels—Hiroki Kubo, NHK - Tokyo, Japan; Satoshi Oode, NHK Science & Technology Research Laboratories - Setagaya-ku, Tokyo, Japan; Takehiro Sugimoto, NHK - Setagaya-ku, Tokyo, Japan; Shu Kitajima, NHK Science & Technology Research Laboratories - Tokyo, Japan; Atsuro Ito, NHK Science & Technology Research Laboratories - Tokyo, Japan; Tomoyasu Komori, NHK Science and Technology Research Laboratories - Setagaya-ku, Tokyo, Japan; Waseda University - Shinjuku-ku, Tokyo, Japan; Kazuho Ono, NHK Science & Technology Research Laboratories - Setagaya-ku, Tokyo, Japan
A unified algorithm to derive downmixing coefficients from a specific multichannel format to arbitrary reproduction formats with a smaller number of channels was investigated. The proposed method attaches importance to maintaining both basic audio quality and spatial impression. It involves appropriately changing the positions of channels in source formats, or locating the channels in source formats at equal intervals between the two channels in destination formats when phantom sources are used. Owing to this feature, this method can minimize the use of phantom sources and avoid the deterioration of spatial impression. A subjective evaluation was carried out and the obtained results implied that the proposed algorithm satisfies our requirements.
Convention Paper 10044
P03-7 Subjective Evaluation of Stereo-9.1 Upmixing Algorithms Using Perceptual Band Allocation—Sungsoo Kim, New York University - New York, NY, USA
The purpose of this study was to investigate preexisting algorithms for building an upmixing algorithm that converts a stereo signal to 5.1 and 9.1 multichannel audio formats. Using three algorithms (the passive surround decoding method, the Least Mean Squares algorithm, the adaptive panning algorithm), a stereo audio signal was upmixed to 5.1 and 9.1 in Max. The Max patch provides a GUI in which listeners can select one of the upmixing algorithms and control EQ during playback. Perceptual Band Allocation (PBA) is applied for converting the upmixed 5.1 channel audio to 9.1 that contains four height channels (top front left, top front right, top back left, top back right). A subjective listening test was conducted in New York University’s MARL Research Lab. LMS algorithm was found to provide more natural and spatial sounds than the other two algorithms. The passive surround decoding method and the adaptive panning algorithm were found to show similar characteristics in terms of low frequency and spaciousness.
Convention Paper 10045