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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   

 

Personal Sound Zones: A Comparison between Frequency and Time Domain Formulations in a Transportation Context

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This paper compares the formulation of a least-squares pressure matching algorithm in the frequency and time domains for the generation of Personal Sound Zones (PSZ) for a transportation application. Due to variations in the transportation’s acoustic environment, the calculation time is added to the usually found metrics in the PSZ bibliography (like Acoustic Contrast, Effort, etc.). Both formulations are implemented to control two zones in three configurations (4, 6, and 8 sources), using monopole simulations and anechoic measurements. In spite of not always presenting perfectly causal filters—pre-ringing in some filters occurs in some cases—the frequency domain formulation allows achieving equal levels of Acoustic Contrast, Effort, and Reproduction error more than 500 times faster than the time domain formulation.

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Mitigating the Effect of In-Vehicle Road Noise Cancellation on Music Playback

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A Road Noise Cancellation (RNC) system is an Active Noise Cancellation (ANC) system implemented in a vehicle in order to minimize undesirable road noise inside the passenger cabin. Current RNC systems undesirably affect the frequency response of music playback. The RNC system’s error microphones sense all the sound in the passenger cabin, including the music. Hence, RNC systems will cancel this total sensed sound and not only the road induced noise. A new True Audio algorithm can directly remove the music signal from the error microphone signals and leave only the interior noise portion. In order to correctly estimate the music portion at the error microphones, True Audio implements a novel control topology based on a new multiple channel, real time modeling of the music’s secondary path transfer function. To validate the effectiveness of the proposed algorithm, experimental and numerical simulations were performed. The numerical studies use logs of real sensors mounted on a vehicle forming an RNC system with six reference accelerometers, five control speakers and six error microphones. Both the models and measurements show that the True Audio algorithm preserves the frequency response of music when the RNC system is activated.

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Effect of a Global Metronome on Ensemble Accuracy in Networked Music Performance

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Several rhythmic experiments with pairs drawn from a group of 23 subjects were performed to investigate the effect of a global metronome on the ensemble accuracy in Networked Music Performance (NMP). Artificial delays up to 91 ms were inserted into the audio transmission between the subjects. To investigate the dependencies between delay times, ensemble accuracy and the highly synchronized global metronome, the experiments were evaluated in terms of tempo acceleration, imprecision and subjective judgment of the ensemble play. The results show that the global metronome leads to a stabilization of the tempo acceleration caused by the delay. The imprecision stays constant to a threshold of about 28 ms and 36 ms, depending on the delay compensating strategy the subjects used. Winner of the 147th AES Convention Student Paper Award

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Evaluation of Multichannel Audio in Automobiles versus Mobile Phones

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Multichannel surround and 3D audio are slowly gaining popularity and eventually commercial content in these formats will become common. Many automobiles still have a stereo sound system with some firmware or software that is capable of rendering multichannel audio into stereo. This paper shows the results of a listening test for multichannel audio conducted in a medium-sized car. The results of this test are compared to the results of a listening test for the same audio excerpts but conducted on a mobile phone with headphones. The results show that on mobile phones, multichannel audio clearly outperforms stereo in terms of perceived audio quality as rated by a user. However in automobiles, multichannel audio only shows marginal improvement in the rated audio quality.

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Realizing an Acoustic Vector Network Analyzer

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Acoustic absorption, reflection, and transmission is typically measured using an impedance tube. We present the design and initial measurements of a radically different measurement system. The instrument builds on the rich history and deep mathematics developed in pursuit of electromagnetic Vector-corrected Network Analyzers (VNAs). Using acoustic directional couplers and a traditional VNA mainframe we assembled an “Acoustic Vector Network Analyzer” (AVNA). The instrument measures acoustic scattering parameters, the complex reflection and transmission coefficients, of materials, transmission lines, ported structures, ducts, etc. After the fashion of electromagnetic VNAs we have constructed millimeter-wave measurement heads that span the 800 Hz–2200 Hz (420–150 mm) and 10 kHz–22 kHz (35–15 mm) bands, demonstrating scalability. We present initial measurement results.

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Analyzing and Extracting Multichannel Sound Field

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Current post production workflow requires sound engineers to create multiple multichannel audio delivery formats. Inaccurate translation between formats may lead to more time and cost for extra manual adjustment; whereas in sound reproduction, it causes misinterpretation of the original mix and deviation from the intended story. This paper proposes a method that combines both analyzing an encoded Ambisonics field from the input multichannel signal and analyzing between each pair of adjacent channels. This allows an overall understanding of the multichannel sound field while having the ability to have a fine extraction from each channel pair. The result can be used to translate between multichannel formats and also to provide a more accurate rendering for immersive stereo playback.

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Profiling Audio Compressors with Deep Neural Networks

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We present a data-driven approach for predicting the behavior of (i.e., profiling) a given parameterized, non-linear time-dependent audio signal processing effect. Our objective is to learn a mapping function that maps the unprocessed audio to the processed, using time-domain samples. We employ a deep auto-encoder model that is conditioned on both time-domain samples and the control parameters of the target audio effect. As a test-case, we focus on the offline profiling of two dynamic range compressors, one software-based and the other analog. Our results show that the primary characteristics of the compressors can be captured, however there is still sufficient audible noise to merit further investigation before such methods are applied to real-world audio processing workflows.

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On the Similarity between Feedback/Loopback Amplitude and Frequency Modulation

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This paper extends previous work in loopback frequency modulation (FM) to a similar system in which an oscillator is looped back to modulate its own amplitude, so called feedback amplitude modulation (FBAM). A continuous-time closed-form solution is presented for each, yielding greatly improved numerical properties, reduced dependency on sampling rate, and a more accurate representation of the feedback by eliminating the unit-sample delay required for discrete-time implementation. Producing similar waveforms, it is shown that FBAM for a known input frequency, is actually a scaled and offset version of loopback FM having a different carrier frequency but same sounding frequency. Two distinct representations are used to show mathematical equivalence between systems while validating the closed-form solution for each.

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Digital Parametric Filters Beyond Nyquist Frequency

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Filter Digitization through the Bilinear Transformation is often considered a very good all-around method to produce equalizer sections. The method is well behaved in terms of stability and ease of implementation; however, the frequency warping produced by the transformation leads to abnormalities near the Nyquist frequency. Moreover, it is impossible to design parametric sections whose analog center frequencies are defined above the Nyquist frequency. These filters, even with center frequencies outside of the hearing range, have effects that extend into the hearing bandwidth with desirable characteristics during mixing and mastering. Surpassing these limitations, while controlling the abnormalities of the warping produced by the Bilinear Transform through an alternative definition of the Bilinear constant is the purpose of this paper. In the process, also a correction factor is discussed for the bandwidth of the parametric section to correct abnormalities affecting the digitization of this parameter.

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Using Volterra Series Modeling Techniques to Classify Black-Box Audio Effects

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Digital models of various audio devices are useful for simulating audio processing effects, but developing good models of nonlinear systems can be challenging. This paper reports on the in-progress work of determining attributes of black-box audio devices using Volterra series modeling techniques. In general, modeling an audio effect requires determination of whether the system is linear or nonlinear, time-invariant or –variant, and whether it has memory. For nonlinear systems, we must determine the degree of nonlinearity of the system, and the required parameters of a suitable model. We explain our work in making educated guesses about the order of nonlinearity in a memoryless system and then discuss the extension to nonlinear systems with memory.

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                 Search Results (Displaying 1-10 of 102 matches)
AES - Audio Engineering Society