Download Faust2android: a Faust Architecture For Android faust2android is a tool that turns a FAUST program into an Android application. Signal processing tasks as well as accessing the audio record and playback resources are done natively in C++ using the Android Native Development Toolkit (NDK). User interface and other components of the application are programmed in JAVA. The implementation as well as issues related to real-time signal processing on Android platforms are discussed. faust2android is part of a larger project whose goal is to build a full FAUST environment for Android: FAUST D ROID.
Download Incremental Functional Reactive Programming for Interactive Music Signal Processing Textual music programming languages offer greater expressive power than diagrammatic visual programming languages and semi-modular graphical user interfaces. However, textual music programming languages don’t allow fine-grained incremental updates to the signal flow graph—instead, they only allow course-grained updates at the statement level. In both the diagrammatic visual programming language and the graphical user interface paradigms, users can directly adjust a parameter by editing the value inline, and such an adjustment does not affect the state of any other part of the signal flow graph. For example, adjusting the attack time of an envelope does not affect the contents of a delay line. By contrast, users of textual music programming languages must either: 1) assign names to nodes and then later use a separate statement to adjust the parameter value, or 2) lose node states (eg. envelope positions, instantaneous LFO phases, and delay line contents) by reevaluating the entire original statement or program. We present a new paradigm in which users can directly edit programs without losing state. In our approach, time-varying programs evaluate to time-varying signal processing graphs, and incremental updates to the program result in incremental updates to the signal processing graph.
Download Music Dereverberation by Spectral Linear Prediction in Live Recordings In this paper, we present our evaluations in using blind single channel dereverberation on music signals. The target material is heavily reverberated and dynamic range compressed polyphonic music from several genres. The applied dereverberation method is based on spectral subtraction regulated by a time-frequency domain linear predictive model. We present our results on enhancing music signal quality and automatic beat tracking accuracy with the proposed dereverberation method. Signal quality enhancement, measured by improvement in signal to distortion ratio, is achieved for both reverberant and dynamic range compressed signals. Moreover, the algorithm shows potential as a preprocessing method for music beat tracking.
Download Audio Time-Scaling for Slow Motion Sports Videos Slow motion videos are frequently featured during broadcast of sports events. However, these videos do not feature any audio channel, apart from the live ambiance and comments from sports presenters. Standard audio time-scaling methods were not developed with such noisy signal in mind and they do not always permit to obtain an acceptable acoustic quality. In this work, we present a new approach that creates high-quality time-stretched version of sport audio recordings while preserving all their transient events.
Download Error Robust Delay-Free Lossy Audio Coding Based on ADPCM We consider the problem of transmission errors in the well known adaptive differential pulse code modulation (ADPCM) system. A single transmission error destabilizes the reconstruction process at the decoder side in the ADPCM coding scheme if a non-leaky algorithm is used. We propose a delay-free and fixed rate of 3 bit/sample audio source coding scheme based on a robust prediction. The prediction of the backward ADPCM coding scheme is attained as a FIR filter in lattice structure. The prediction filter is derived as a reconstructed-signal-driven (RSD) or a predictionerror-driven (PED) lattice filter. A technique for an error robust RSD prediction is presented. This technique is employed in a robust audio coding scheme without use of any additional overhead. The proposed modified RSD-ADPCM is compared to the PED-ADPCM coding scheme by means of the objective audio quality. The proposed system yields good objective audio quality in the noise-free channels and provides robustness in the presence of transmission errors.
Download Selection And Interpolation of Head-Related Transfer Functions for Rendering Moving Virtual Sound Sources A variety of approaches have been proposed previously to interpolate head-related transfer functions (HRTFs). However, relatively little attention has been given to the way a suitable set of HRTFs is chosen for interpolation and to the calculation of the interpolation weights. This paper presents an efficient and robust way to select a minimal set of HRTFs and to calculate appropriate weights for interpolation. The proposed method is based on grouping HRTF measurement points into non-overlapping triangles on the surface of a sphere by calculating the convex hull. The resulting Delaunay triangulation maximises minimum angles. For interpolation, the HRTF triangle that is intersected by the desired sound source vector is selected. The selection is based on a point-in-triangle test than can be performed using just 9 multiplications and 6 additions per triangle. A further improvement of the selection process is achieved by sorting the HRTF triangles according to their distance from the sound source vector prior to performing the pointin-triangle tests. The HRTFs of the selected triangle are interpolated using weights derived from vector-base amplitude panning, with appropriate normalisation. The proposed method is compared to state-of-the-art methods. It is shown to be robust with respect to irregularities in the HRTF measurement grid and to be well-suited for rendering moving virtual sources.
Download Room Acoustics Modelling using Gpu-Accelerated Finite Difference and Finite Volume Methods On a Face-Centered Cubic Grid In this paper, a room acoustics simulation using a finite difference approximation on a face-centered cubic (FCC) grid with finite volume impedance boundary conditions is presented. The finite difference scheme is accelerated on an Nvidia Tesla K20 graphics processing unit (GPU) using the CUDA programming language. A performance comparison is made between 27-point finite difference schemes on a cubic grid and the 13-point scheme on the FCC grid. It is shown that the FCC scheme runs faster on the Tesla K20 GPU and has less numerical dispersion than best 27-point schemes on the cubic grid. Implementation details are discussed.
Download A New Reverberator based on Variable Sparsity Convolution An efficient algorithm approximating the late part of room reverberation is proposed. The algorithm partitions the impulse response tail into variable-length segments and replaces them with a set of sparse FIR filters and lowpass filters, cascaded with several Schroeder allpass filters. The sparse FIR filter coefficients are selected from a velvet noise sequence, which consists of ones, minus ones, and zeros only. In this application, it is sufficient perceptually to use very sparse velvet noise sequences having only about 0.1 to 0.2% non-zero elements, with increasing sparsity along the impulse response. The algorithm yields a parametric approximation of the late part of the impulse response, which is more than 100 times more efficient computationally than the direct convolution. The computational load of the proposed algorithm is comparable to that of FFT-based partitioned convolution techniques, but with nearly half the memory usage. The main advantage of the new reverberator is the flexible parameterization.
Download B-Format Acoustic Impulse Response Measurement and Analysis In the Forest at Koli National Park, Finland Acoustic impulse responses are used for convolution based auralisation and reverberation techniques for a range of applications, such as music production, sound design and virtual reality systems. These impulse responses can be measured in real world environments to provide realistic and natural sounding reverberation effects. Analysis of this data can also provide useful information about the acoustic characteristics of a particular space. Currently, impulse responses recorded in outdoor conditions are not widely available for surround sound auralisation and research purposes. This work presents results from a recent acoustic survey of measurements at three locations in the snow covered forest of Koli National Park in Finland during early spring. Acoustic impulse responses were measured using a B-format Soundfield microphone and a single loudspeaker. The results are analysed in terms of reverberation and spatial characteristics. The work is part of a larger study to collect and investigate acoustic impulse responses from a variety of outdoor locations under different climatic conditions.
Download A Scalable Architecture for General Real-Time Array-Based DSP on FPGAs with Application to the Wave Equation This paper describes a scheme for parallel execution on FPGAs of DSP tasks which rely heavily on MAC operations. Multiple operations are assigned to a single ‘processing node’ such that each node can operate just in real-time. Where the number of MACs required exceeds the capability of a single processing node additional nodes are added until the capacity of the FPGA is exhausted. Additional requirements beyond the capability of a single FPGA are accommodated by extension across multiple devices, offering significant scalability. Resource usage, performance results for an example acoustic modelling application on a modest single FPGA and development system are presented.