Download Sub-Band Independent Subspace Analysis for Drum Transcription
While Independent Subspace Analysis provides a means of separating sound sources from a single channel signal, making it an effective tool for drum transcription, it does have a number of problems. Not least of these is that the amount of information required to allow separation of sound sources varies from signal to signal. To overcome this indeterminacy and improve the robustness of transcription an extension of Independent Subspace Analysis to include sub-band processing is proposed. The use of this approach is demonstrated by its application in a simple drum transcription algorithm.
Download An Extension for Source Separation Techniques Avoiding Beats
The problem of separating individual sound sources from a mixture of these, known as Source Separation or Computational Auditory Scene Analysis (CASA), has become popular in the recent decades. A number of methods have emerged from the study of this problem, some of which perform very well for certain types of audio sources, e.g. speech. For separation of instruments in music, there are several shortcomings. In general when instruments play together they are not independent of each other. More specifically the time-frequency distributions of the different sources will overlap. Harmonic instruments in particular have high probability of overlapping partials. If these overlapping partials are not separated properly, the separated signals will have a different sensation of roughness, and the separation quality degrades. In this paper we present a method to separate overlapping partials in stereo signals. This method looks at the shapes of partial envelopes, and uses minimization of the difference between such shapes in order to demix overlapping partials. The method can be applied to enhance existing methods for source separation, e.g. blind source separation techniques, model based techniques, and spatial separation techniques. We also discuss other simpler methods that can work with mono signals.
Download Real Time Implementation of the HVXC MPEG-4 Speech Coder
In this paper we present the results of the code optimization for the HVXC MPEG-4 speech coder. Two kinds of bit-rate formats are considered: 2 and 4 kbit/s. After a short description of the HVXC main features, results of code optimization are reported: the real time implementationon, on a floating point DSP, of three parallel 2 kbit/s or two parallel 4 kbit/s HVXC coders, is shown to be possible.
Download Optimizing Digital Musical Effect Implementation for Multiple Processor DSP Systems
In the area of digital musical effect implementation, attention has lately been focused on computer workstations designed for digital processing of sound, which perform all operations with audio signals in real time. They are in fact a combination of powerful computer program and hardware cards with digital signal processors. Thanks to the power enhancement of personal computer core, performing these operations in the CPU is currently possible. However, in most cases, digital signal processors are still used for these purposes because digital musical effect modelling is more effective and more precise with the digital signal processor. In addition to this, processing in digital signal processor saves the CPU computing power for other functions.
Download Realization of a Diffuse Sound Field with a PC-Based Sound Card Solution
For the quality assessment of headphones, especially the loudness measuring of headphones, a diffuse sound field is required. At this time a hardware based noise generator, one-third octave filters built up in analog mode as well as boosters are used. In this work a flexible PC-based solution with the aid of a sound card is presented. Therefore ten independent noise generators, generating Gaussian distributed white noise, are needed. The implementation using the ’Dynamic Creation of Pseudorandom Number Genrators’ for ’Mersenne Twister’ is described. A probability transformation to convert equal distributed numbers into Gaussian distributed ones is derived in detail. Furthermore one-third octave filters are designed and implemented according to the ANSI standard. The access to the sound card is provided using the Wave-API library under Microsoft Windows. This work was carried out at Sennheiser electronic GmbH in Wennebostel (Germany) in the development department for cord based headphones.
Download A new Scheme for Real-Time Loop Music Production Based on Granular Similarity and Probability Control
In this paper, a new concept of real-time loop music production is introduced, along with its implementation in Pd. This scheme tends to improvise loop music based on very limited pattern (loop sample) materials. Four loops, each divided into 32 grains, work at the same time. Analysis of the spectral and energy similarities between every two grains are conducted, and the transition probability matrices generated by the analysis phase are consulted for each decision of grain choice during remixing. A joystick-style controller is designed to control the probability distribution, which changes the music characteristics in real-time. While maintaining some characteristics of each loop pattern, the music generated by the program reveals a large space of variations and controllable improvisations. Real-time analysis is considered, that later will enable switching new patterns into the 4pattern group during a performance. This scheme is a potential new method for live computer DJ mixing in the loop pattern level. Sound examples, including four drum loops and the improvisations on them, can be found at˜pxiang/granuloop.htm
Download Real-Time Partial Tracking in an Augmented Additive Synthesis System
This paper describes an approach to real time partial tracking in an analysis\transformation\resynthesis system using a combination of linear and bi-linear time-frequency techniques. Tests of the system have been made using both natural and synthetic sounds. Results are presented and areas for further research and development are discussed.
Download Efficient Parametric Modeling for Audio Transients
In this work, we present an evolution of the DDS (Damped & Delayed Sinusoidal) model introduced within the framework of the general signal modeling. This model is named Partial Damped & Delayed Sinusoidal (PDDS) model and takes into account a single time delay parameter for a set of (un)damped sinusoids. This modification is more consistent with the transient audio modeling problem. Then, we develop model parameter high-resolution estimation algorithms. Simulations on a typical transient audio signals show the validity of this approach.
Download Adapting the Overlap-Add Method to the Synthesis of Noise
Spectral synthesis techniques often use the OverLap-Add method. But in the case of noise synthesis, both experiments and theory show that this method introduces intensity fluctuations which imply audible artifacts. We propose here new methods to avoid these variations. The first one consists in multiplying the resulting signal by another weighting window to compensate dynamic fluctuations. The second one defines a new OLA weighting window. The third one concerns only noises synthesized with sinusoidal components and uses time-shifting to cancel artifacts.
Download Audio Signal Extrapolation - Theory and Applications
A method for extrapolating discrete audio signals is described. The theory of extrapolation is studied and some applications are presented and demonstrated. The extrapolation method is fast and capable of extrapolating several thousand samples of CD-quality audio signals. The extrapolation is applied in practice to enhance the spectral resolution in short-time fast Fourier transform based methods. It is also applied to eliminate impulsive noise bursts and to recover missing signal sections.