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In this paper we propose a long-term prediction method for low delay transform domain general audio coders. This Frequency Domain Joint Harmonics Prediction (FDJHP) method operates directly in the Modified Discrete Cosine Transform (MDCT) domain and can enhance the coding efficiency, even under very low frequency resolutions. We compare this new method with state-of-the-art MDCT based methods by analyzing bitrate savings and by a listening test using test signals with strong harmonic components. The results indicate that it outperforms an existing method, which also directly operates in the frequency domain. Additionally, we show how it can be combined with the existing techniques into an adaptive system, where the different methods can complement each other.

Speech enhancement involves the distinction of a target speech signal from an intrusive background. Although generative approaches using Variational Autoencoders or Generative Adversarial Networks (GANs) have increasingly been used in recent years, normalizing flow (NF) based systems are still scarse, despite their success in related fields. Thus, in this paper we propose a NF framework to directly model the enhancement process by density estimation of clean speech utterances conditioned on their noisy counterpart. The WaveGlow model from speech synthesis is adapted to enable direct enhancement of noisy utterances in time domain. In addition, we demonstrate that nonlinear input companding benefits the model performance by equalizing the distribution of input samples. Experimental evaluation on a publicly available dataset shows comparable results to current state-of-the-art GAN-based approaches, while surpassing the chosen baselines using objective evaluation metrics., Comment: Accepted to ICASSP 2021

Perceptual audio coding schemes typically apply the modified discrete cosine transform (MDCT) with different lengths and windows, and utilize signal-adaptive switching between these on a perframe basis for best subjective performance. In previous papers, the authors demonstrated that further quality gains can be achieved for some input signals using additional transform kernels such as the modified discrete sine transform (MDST) or greater inter-transform overlap by means of a modified extended lapped transform (MELT). This work discusses the algorithmic procedures and codec modifications necessary to combine all of the above features—transform length, window shape, transform kernel, and overlap ratio switching—into a flexible input-adaptive coding system. It is shown that, due to full time-domain aliasing cancelation, this system supports perfect signal reconstruction in the absence of quantization and, thanks to fast realizations of all transforms, increases the codec complexity only negligibly. The results of a 5.1 multichannel listening test are also reported.

Time Domain Aliasing Reduction (TDAR) is a method to improve the impulse response compactness of non-uniform orthogonal Modified Discrete Cosine Transforms (MDCT). Previously, TDAR was only possible between frames of identical time-frequency tilings, however in this letter we describe a method to overcome this limitation. This method enables the use of TDAR between two consecutive frames of different time-frequency tilings by introducing another subband merging or subband splitting step. Consecutively, this method allows more flexible and adaptive filterbank tilings while retaining compact impulse responses, two attributes needed for efficient perceptual audio coding.

In this letter we describe nonuniform orthogonal modified discrete cosine transform (MDCT) filterbanks and time-domain aliasing reduction (TDAR). By adding a postprocessing step to the MDCT, our method allows for arbitrary nonuniform frequency resolutions using subband merging with smooth windowing and overlap in frequency. This overlap allows for an improved temporal compactness of the impulse response, which is especially useful for audio coders. The postprocessing step comprises another lapped MDCT transform along the frequency axis and TDAR along each subband signal.

During December 1995, subjective tests were carried out by members of the Moving Picture Experts Group (MPEG, ISO/JTC1/SC29/WG11) to select the proposed technology for inclusion in the audio part of the new MPEG-4 standard. The new standard addresses coding for more than just the functionality of data rate compression. Material coded at very low bit-rates is also included. Thus, different testing methodologies were applied, according to ITU-R Rec. BS 1116 for a bit-rate of 64 kbit/s per channel and according to ITU-T Rec. P.80 for lower bit-rates or functionalities other than data rate compression. Proposals were subjectively tested for coding efficiency, error resilience, scalability and speed change: a subset of the MPEG-4 ‘functionalities’. This paper describes how two different evaluation methods were used and adjusted to fit the different testing requirements. This first major effort to test coding schemes at low bit-rates proved successful. Based on the test results, decisions for MPEG-4 technology were made. This was the first opportunity for MPEG members to carry out tests on the submitted functionalities. In the process, much was learnt. As a result, some suggestions are made to improve the way new functionalities can be subjectively evaluated.

A common theory of lapped orthogonal transforms (LOTs) and critically sampled filter banks, called L into N coding (LINC), is presented. The theory includes a unified analysis of both coding methods and identity relations between the transform, inverse transform, analysis filter bank, and synthesis filter bank. A design procedure for LINC analysis/synthesis systems, which satisfy the conditions for perfect reconstruction, is developed. The common LINC theory is used to define an ideal LINC system which is used, together with the power spectral density of the input signal, to calculate theoretical bounds for the coding gain. A generalized overlapping block transform (OBT) with time domain aliasing cancellation (TDAC) is used to approximate the ideal LINC. A generalization of the OBT includes multiple block overlap and additional windowing. A recursive design procedure for windows of arbitrary lengths is presented. The coding gain of the generalized OBT is higher than that of the Karhunen-Loeve transform (KLT) and close to the theoretical bounds for LINC. In the case of image coding, the generalized OBT reduces the blocking effects when compared with the DCT. >

The decomposition of audio signals into perceptually meaningful multiband modulation components opens up new possibilities for advanced signal processing. The signal adaptive analysis approach proposed in this paper will be shown to provide a powerful handle on the signal's perceptual properties: pitch, timbre or roughness can be manipulated straight forward. Additionally a synthesis method is specified providing high subjective perceptual quality. Furthermore, as an application example, a novel audio processing technique is proposed which changes the key mode of a given piece of music e.g. from major to minor key or vice versa.

Today, cochlear implants (CIs) are the treatment of choice in patients with profound hearing loss. However speech intelligibility with these devices is still limited. A factor that determines hearing performance is the processing method used in CIs. Therefore, research is focused on designing different speech processing methods. The evaluation of these strategies is subject to variability as it is usually performed with cochlear implant recipients. Hence, an objective method for the evaluation would give more robustness compared to the tests performed with CI patients. This paper proposes a method to evaluate signal processing strategies for CIs based on a hidden markov model speech recognizer. Two signal processing strategies for CIs, the Advanced Combinational Encoder (ACE) and the Psychoacoustic Advanced Combinational Encoder (PACE), have been compared in a phoneme recognition task. Results show that PACE obtained higher recognition scores than ACE as found with CI recipients.

A physiological and computational model of the human auditory system has been fitted in a signal processing strategy for cochlear implants (CIs). The aim of the new strategy is to obtain more natural sound in CIs by better mimicking the human auditory system. The new strategy was built in three independent stages as proposed in [6]. First a basilar membrane motion model was substituted by the filterbank commonly used in commercial strategies. Second, an inner hair cell model was included in a commercial strategy while maintaining the original filterbank. Third, both the basilar membrane motion and the inner-hair cell model were included in the commercial strategy. This paper analyses the properties and presents results obtained with CI recipients for each algorithm designed.

Current speech processing in cochlear implants use a filterbank to analyse audio signals into several frequency bands, each associated with one electrode. Because the processing is performed on input signal blocks of fixed sizes, the filterbank provides a unique time-frequency resolution to represent the various signal features. However, different components of audio signals may require different time-frequency resolutions for an accurate representation and perception. In this paper we investigate the influence on speech intelligibility in cochlear implants users when filterbanks with different time-frequency resolutions are used. In order to represent all signal features accurately, an adaptive filterbank has been developed that accepts input blocks of different sizes. The different resolutions required are achieved by adequately switching between block sizes depending on the input signal characteristics. The filterbank was incorporated into the commercial Advanced Combinational Encoder (ACE) and acutely tested on six cochlear implant recipients.

Current speech processing strategies for cochlear implants use a filterbank which decomposes the audio signals into multiple frequency bands each associated with one electrode. Pitch perception with cochlear implants is related to the number of electrodes inserted in the cochlea and to the rate of stimulation of these electrodes. The filterbank should, therefore, be able to analyze the time-frequency features of the audio signals while also exploiting the time-frequency features of the implant. This study investigates the influence on speech intelligibility in cochlear implant users when filterbanks with different time-frequency resolutions are used. Three filter-banks, based on the structure of a wavelet packet transform but using different basis functions, were designed. The filter-banks were incorporated into a commercial speech processing strategy and were tested on device users in an acute study.

We describe a new signal processing technique for cochlear implants using a psychoacoustic-masking model. The technique is based on the principle of a so-called "NofM" strategy. These strategies stimulate fewer channels ( ) per cycle than active electrodes (NofM; ). In "NofM" strategies such as ACE or SPEAK, only the channels with higher amplitudes are stimulated. The new strategy is based on the ACE strategy but uses a psychoacoustic-masking model in order to determine the essential components of any given audio signal. This new strategy was tested on device users in an acute study, with either 4 or 8 channels stimulated per cycle. For the first condition (4 channels), the mean improvement over the ACE strategy was . For the second condition (8 channels), no significant difference was found between the two strategies.

In contemporary cochlear implant systems, the audio signal is decomposed into different frequency bands, each assigned to one electrode. Thus, pitch perception is limited by the number of physical electrodes implanted into the cochlea and by the wide bandwidth assigned to each electrode. The Harmony HiResolution bionic ear (Advanced Bionics LLC, Valencia, CA, USA) has the capability of creating virtual spectral channels through simultaneous delivery of current to pairs of adjacent electrodes. By steering the locus of stimulation to sites between the electrodes, additional pitch percepts can be generated. Two new sound processing strategies based on current steering have been designed, SpecRes and SineEx. In a chronic trial, speech intelligibility, pitch perception, and subjective appreciation of sound were compared between the two current steering strategies and standard HiRes strategy in 9 adult Harmony users. There was considerable variability in benefit, and the mean results show similar performance with all three strategies.

Das hier vorgestellte Verfahren der Bitratenreduktion für Audiosignale basiert auf überlappenden Transformationen mit „Time Domain Aliasing Cancellation“, deren Fensterfunktionen und Transformationslängen in Abhängigkeit vom Eingangssignal umgeschaltet werden. Die adaptive Fensterung verbessert das Verhalten der Transformationscodierung mit überlappenden Blöcken, die sich durch einen hohen Codierungsgewinn auszeichnet, beim Auftreten von Impulsen und Amplitudensprüngen im Eingangssignal. © 1989, Walter de Gruyter. Alle Rechte vorbehalten.