Bulk download - click topic to download Zip archive of all papers related to that topic: Applications in Audio Audio Education Audio Signal Processing Perception Posters: Applications in Audio Posters: Audio Signal Processing Posters: Perception Posters: Recording and Production Posters: Room Acoustics Posters: Spatial Audio Posters: Transducers Product Development Recording and Production Recording, Production, and Live Sound Room Acoustics Semantic Audio Spatial Audio Spatial Audio, Part 1 Spatial Audio, Part 2 Spatial Audio, Part 3 Transducers
The authors present a novel approach to recording classical music, making use of non-adjacent 3rd order Ambisonics microphone arrays. The flexible combination of source separated signals with varied degrees of beamforming focus enable independent levels control, while maintaining the spatial coherence and reverberation qualities of the recorded spaces. The non-coincidental arriving locations of multiple arrays allow for post-production manipulations without disrupting the inherent classical music logic that values the overall sound as opposed to individual single sound sources. In addition, this method employs portable and lightweight equipment to record decorrelated signals, which can be mixed in surround formats with enhanced sense of depth.
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We investigate listener preference in multitrack music production using the Mix Evaluation Dataset, comprised of 184 mixes across 19 songs. Features are extracted from verses and choruses of stereo mixdowns. Each observation is associated with an average listener preference rating and standard deviation of preference ratings. Principal component analysis is performed to analyze how mixes vary within the feature space. We demonstrate that virtually no correlation is found between the embedded features and either average preference or standard deviation of preference. We instead propose using principal component projections as a semantic embedding space by associating each observation with listener comments from the Mix Evaluation Dataset. Initial results disagree with simple descriptions such as “width” or “loudness” for principal component axes.
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There is a body of work in the field of intelligent music production covering a range of specific audio effects. However, there is a distinct lack of any purely machine learning approaches to automatic mixing. This could be due to a lack of suitable data. This paper presents an approach to used human produced audio mixes, along with their source multitrack, to produce the set of mix parameters. The focus will be entirely on the gain mixing of audio drum tracks. Using existing reverse engineering of music production gain parameters, a target mix gain parameter is identified, and these results are fed into a number of machine learning algorithms, along with audio feature vectors of each audio track. This allow for a machine learning prediction approach to audio gain mixing. A random forest approach is taken to perform a multiple output prediction. The prediction results of the random forest approach are then compared to a number of other published automatic gain mixing approaches. The results demonstrate that the random forest gain mixing approach performs similarly to that of a human engineer and outperforms the existing gain mixing approaches.
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The New York University and the Leibniz University Hannover are working on future immersive Networked Music Performances. One of the biggest challenges of audio data transmission over IP-based networks is latency, which can affect the interplay of the participants. In this contribution, two metronomes, utilizing the Global Positioning System to generate a globally synchronized click signal, were used as a tool to determine delay times in the data transmission between both universities with high precision. The aim of this ?rst study is to validate the proposed method by obtaining insights into transmission latency as well as latency ?uctuations and asymmetries. This work also serves as baseline for future studies and helps to establish an effective connection between the two institutions.
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Loudspeaker-room interactions are well known for affecting the perceived sound quality of low frequencies. To solve this problem, different room adaptation systems for adapting a loudspeaker to its acoustic environment have been developed. In this study two listening tests were performed to assess the effectiveness of four different room adaptation systems under different circumstances. The factors investigated include the listening room, loudspeaker, listening position, and listener. The results indicate that listeners’ preference for different adaptation systems is affected by the specific acoustic environment. It was found that the adaptation system based on acoustic power measurement proved to be more preferred, also with stable performance.
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The sine sweep is one of the most effective methods for measuring room impulse responses; however, ambient room noise or unpredictable impulsive noises can negatively affect the quality of the measurement. This study evaluates four different variants of sine sweeps techniques for their resilience to noise when used as an excitation signal in room impulse response measurements: linear, exponential, noise whitened, and minimum noise. The result shows that in a pseudo-anechoic environment, exponential and linear sine sweeps are most resilient to impulsive noise among the four sweeps, while none of the evaluated sine sweeps are resilient to impulsive noise in an acoustically untreated room. Additionally, it is shown that minimum noise sine sweeps are most resilient to ambient noise.
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An approach to find the cause of the perceptual sound quality change by headphone cable has been proposed. This is a method of verifying the validity of the selected candidate by selecting candidate factors from the measurement results, simulating them by digital signal processing, and evaluating the simulated sounds by audition. In the headphone cable, it was found that the factor is that the inductance changes due to the flowing current. It has become clear from the experimental results that changes in transfer characteristics are very sensitively affecting the perceptual sound quality.
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Speech intelligibility/speech clarity is important in any setting in which information is verbally communicated. More specifically, a high level of speech intelligibility is crucial in classrooms to allow teachers to effectively communicate with their students. Given the importance of speech intelligibility in learning environments, several studies have analyzed how accurately the standard method of measuring clarity predicts the level of speech intelligibility in a room. In the context of speech measurements, C50 has been widely used to measure clarity. Instead of using a standard omnidirectional microphone to record room impulse responses for clarity measurements, this study examines the effectiveness of room impulse responses measured with a binaural dummy head. The data collected for this experiment show that C50 measurements differ between the left and right channels by varying amounts based on the dummy head’s position in the room and head orientation. To further investigate the effectiveness of binaural C50 measurements in comparison to the effectiveness of omnidirectional C50 measurements, this research explores the results of psychoacoustic testing to determine which recording method more consistently predicts human speech intelligibility. These results, combined with qualitative observations, predict how precisely acousticians are able to measure C50.
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Inertial exciters are used to actuate a surface into bending vibration, producing sound, but often have a high-Q resonance that can cause the exciter magnet to displace enough to contact the bending panel. The magnet contacting the panel can cause distortion and possibly even damage to the exciter or panel while having a minimal contribution to acoustic output. A method is outlined for deriving a digital biquad filter to cancel out the excessive displacement of the magnet based on measurements of the exciter’s resonant frequency and Q-factor. Measurements of exciter and panel displacement demonstrate that an applied filter reduces magnet excursion by 20 dB at the resonant frequency.
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Music is usually recorded using traditional microphone techniques. With technology continually advancing, binaural recording has become more popular, that is, a recording where two microphones are used to create a three-dimensional stereo image. Commercially available binaural heads are prohibitively expensive and not practical for use in typical educational environments or for casual use in a home studio. This experiment consisted of gathering recorded stimuli with a homemade binaural head and the Neumann KU 100. The recordings were played back for 34 subjects instructed to rate the level of externalization for each example. The study investigates whether a homemade binaural head made for under $500 can externalize sound as well as a commercially available binaural head the Neumann KU 100.
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