For a long time, many popular listening test methods, such as ITU-R BS.1534 (MUSHRA), could not be carried out as web-based listening tests, since established web standards did not support all ...required audio processing features. With the standardization of the Web Audio API, the required features became available and, therefore, also the possibility to implement a wide range of established methods as web-based listening tests. In order to simplify the implementation of MUSHRA listening tests, the development of webMUSHRA was started. By utilizing webMUSHRA, experimenters can configure web-based MUSHRA listening tests without the need of web programming expertise. Today, webMUSHRA supports many more listening test methods, such as ITU-R BS.1116 and forced-choice procedures. Moreover, webMUSHRA is highly customizable and has been used in many auditory studies for different purposes.Keywords: web-based, listening tests, auditory experiments, MUSHRA, Web Audio API
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.
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.
A blind bandwidth extension (BBWE) expands the bandwidth of telephone speech which often is limited to 0.2 to 3.4 kHz. The advantage is an increased perceived quality as well as an increased ...intelligibility. This work presents a BBWE similar to state-of-the-art bandwidth extensions like Intelligent Gap Filling with the difference that all processing is done in the decoder without the need of transmitting extra bits. Parameters like spectral envelope are estimated by a regressive Convolutional Deep Neuronal Network (CNN) with long short-term memory (LSTM). The system operates on frames of 20 ms without additional algorithmic delay and can be applied in state-of-the-art speech and audio codecs.
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.
A common problem when coding transient signals in transform audio codecs is the temporal unmasking of quantization noise leading to artifacts such as pre-echoes. In this work, a new method for ...temporal shaping of quantization noise is introduced. Subband merging/splitting with overlapping windows is applied on the MDCT (Modified Discrete Cosine Transform) spectrum in the encoder and decoder. This allows direct modification of the temporal subband signals for different frequency ranges in a way that results in a flattening of the temporal envelopes in the encoder before quantization and a reshaping to their original envelope at the decoder. The reshaping process at the decoder not only restores the envelopes of the original subband signals, but also shapes the quantization noise accordingly, which results in quantization noise temporally shaped similarly to the original subband signal. This avoids temporal unmasking of quantization noise and the pre-echo artifact. For the temporal flattening/reshaping process the application of gains transmitted via Autoregressive model as well as a low side information approach via companding are proposed and evaluated regarding their noise shaping capabilities in an experimental codec system.
In this paper, we investigate the coding efficiency of perceptual coding using an adaptive non-uniform orthogonal filter-bank based on MDCT analysis/synthesis and time domain aliasing reduction. We ...compare its performance to a system using a traditional adaptive uniform MDCT filterbank with window switching. The comparison is performed using a listening test at two different quantization settings. The statistical evaluation shows that the percetpual quality of the nonuniform filterbank significantly out-performs that of the uniform filterbank by 5 to 10 MUSHRA points.
Efficient combinations of coding and manipulation of audio signals in the spectral domain are often desirable in communication systems. The modified discrete cosine transform (MDCT) represents a ...popular spectral transform in audio coding as it leads to compact signal representations. However, as the MDCT corresponds to a critically sampled filter bank, it is in general not appropriate to directly apply it to filtering tasks. In this paper we present a method to compensate for aliasing terms that arise from such direct MDCT domain filtering. The discussion is thereby based on a rigorous matrix representation of critically sampled filter banks which also leads to corresponding efficient realizations. As an application showcase, noise reduction for MDCT based speech coding is considered in simulations.
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