Toni Hirvonen

Localization of virtual sources in multichannel audio reproduction

The localization of virtual sources generated with different two-dimensional (2-D) multichannel reproduction systems has been studied by means of auditory model simulations and listening tests. The reproduction was implemented with typical five- and eight-channel loudspeaker setups. The microphone systems used were first- and second-order Ambisonics as well as a spaced microphone technique. Pair-wise panning was also studied. The results show that the auditory model can be used in the prediction of perceived direction in multichannel sound reproduction near the median plane. Some systematic deviations between the model predictions and the listening test results were found farther from the median plane. The frequency-dependent capability to produce narrow-band virtual sources to targeted directions is reported for the studied systems.

Single-Channel and Multi-Channel Sinusoidal Audio Coding Using Compressed Sensing

Compressed sensing (CS) samples signals at a much lower rate than the Nyquist rate if they are sparse in some basis. In this paper, the CS methodology is applied to sinusoidally modeled audio signals. As this model is sparse by definition in the frequency domain (being equal to the sum of a small number of sinusoids), we investigate whether CS can be used to encode audio signals at low bitrates. In contrast to encoding the sinusoidal parameters (amplitude, frequency, phase) as current state-of-the-art methods do, we propose encoding few randomly selected samples of the time-domain description of the sinusoidal component (per signal segment). The potential of applying compressed sensing both to single-channel and multi-channel audio coding is examined. The listening test results are encouraging, indicating that the proposed approach can achieve comparable performance to that of state-of-the-art methods. Given that CS can lead to novel coding systems where the sampling and compression operations are combined into one low-complexity step, the proposed methodology can be considered as an important step towards applying the CS framework to audio coding applications.

Functional Count-Comparison Model for Binaural Decoding

Some recent neurophysical studies suggest that mammalian binaural decoding is based on count comparison. When a signal is presented earlier or with higher level to one ear, the neural signals are stronger in the auditory pathways leading to the contralateral hemisphere in such mechanisms. This paper describes functional count-comparison models of two brainstem nuclei, medial superior olive (MSO) and lateral superior olive (LSO), both of which exist in both hemispheres. The topology of the organs and the connections between them as presented in the current neuroanatomical studies are imitated in the functional model. The parameters of the functional models are selected to fit existing neurophysiological and psychoacoustical data. It is shown that the proposed MSO and LSO models are sensitive to interaural differences in time and level in a way that accounts for some known psychoacoustical phenomena.

Interaural Coherence Estimation with Instantaneous ILD

This paper presents a novel computational auditory model inspired by psycho acoustic and neurophysiological findings. The model utilizes the instantaneous difference between the two ear signals to predict spatial hearing cues that humans perceive. The main focus is on the interaural coherence cue, i.e. the perceived similarity between the waveforms of the ear signals. Simulations show that the proposed model is capable of predicting known psychoacoustical results

Decorrelation for audio object coding

Object-based representations of audio content are increasingly used in entertainment systems to deliver immersive and personalized experiences. Efficient storage and transmission of such content can be achieved by joint object coding algorithms that convey a reduced number of downmix signals together with parametric side information that enables object reconstruction in the decoder. This paper presents an approach to improve the performance of joint object coding by adding one or more decorrelators to the decoding process. Listening test results illustrate the performance as a function of the number of decorrelators. The method is adopted as part of the Dolby AC-4 system standardized by ETSI.

Delivering Scalable Audio Experiences using AC-4

AC-4 is a state-of-the-art audio codec standardized in ETSI (TS 103 190 and TS 103 190-2) and included in the DVB toolbox (TS 101 154 V2.2.1 and DVB BlueBook A157) and, at the time of writing, is a candidate standard for ATSC 3.0 as per A/342 part 2. AC-4 is an audio codec designed to address the current and future needs of video and audio entertainment services, including broadcast and Internet streaming. As such, it incorporates a number of features beyond the traditional audio coding algorithms, such as capabilities to support immersive and personalized audio, support for advanced loudness management, video-frame synchronous coding, dialog enhancement, etc. This paper will outline the thinking behind the design of the AC-4 codec, explain the different coding tools used, the systemic features included, and give an overview of performance and applications. It further outlines metadata aspects (immersive and personalized, essential for broadcast), metadata carriage, aspects of interchange of immersive programing, as well as immersive playback and rendering.