Download Feature design for the classification of audio effect units by input/output measurements Virtual analog modeling is an important field of digital audio signal processing. It allows to recreate the tonal characteristics of real-world sound sources or to impress the specific sound of a certain analog device upon a digital signal on a software basis. Automatic virtual analog modeling using black-box system identification based on input/output (I/O) measurements is an emerging approach, which can be greatly enhanced by specific pre-processing methods suggesting the best-fitting model to be optimized in the actual identification process. In this work, several features based on specific test signals are presented allowing to categorize instrument effect units into classes of effects, like distortion, compression, modulations and similar categories. The categorization of analog effect units is especially challenging due to the wide variety of these effects. For each device, I/O measurements are performed and a set of features is calculated to allow the classification. The features are computed for several effect units to evaluate their applicability using a basic classifier based on pattern matching.
Download Low-delay vector-quantized subband ADPCM coding Several modern applications require audio encoders featuring low data rate and lowest delays. In terms of delay, Adaptive Differential Pulse Code Modulation (ADPCM) encoders are advantageous compared to block-based codecs due to their instantaneous output and therefore preferred in time-critical applications. If the the audio signal transport is done block-wise anyways, as in Audio over IP (AoIP) scenarios, additional advantages can be expected from block-wise coding. In this study, a generalized subband ADPCM concept using vector quantization with multiple realizations and configurations is shown. Additionally, a way of optimizing the codec parameters is derived. The results show that for the cost of small algorithmic delays the data rate of ADPCM can be significantly reduced while obtaining a similar or slightly increased perceptual quality. The largest algorithmic delay of about 1 ms at 44.1 kHz is still smaller than the ones of well-known low-delay codecs.
Download Cascaded prediction in ADPCM codec structures The aim of this study is to demonstrate how ADPCM-based codec structures can be improved using cascaded prediction. The advantage of predictor cascades is to allow the adaption to several signal conditions, as it is done in block-based perceptual codecs like MP3, AAC, etc. In other words, additional predictors with a small order are supposed to enhance the prediction of non-stationary signals. The predictor cascade is complemented with a simple adaptive quantizer to yield a simple exemplary codec which is used to demonstrate the influence of the predictor cascade. Several cascade configurations are considered and optimized using a genetic algorithm. A measurement of the prediction gain and the ODG score utilizing the PEAQ algorithm applied to the SQAM dataset shall reveal the potential improvements.
Download Downmix compatible conversion from mono to stereo in time- and frequency-domain Even in a time of surround and 3D sound, many tracks and recordings are still only available in mono or it is not feasible to record a source with multiple microphones for several reasons. In these cases, a pseudo stereo conversion of mono signals can be a useful preprocessing step and/or an enhancing audio effect. The conversion proposed in this paper is designed to deliver a neutral sounding stereo image by avoiding timbral coloration or reverberation. Additionally, the resulting stereo signal is downmix-compatible and allows to revert to the original mono signal by a simple summation of the left and right channels. Several configuration parameters are shown to control the stereo panorama. The algorithm can be implemented in time-domain or also in the frequency-domain with additional features, like center focusing.