APPLICATIONS OF DIRECTIONAL AUDIO CODING IN AUDIO

Directional Audio Coding (DirAC) is a method for spatial sound representation, applicable to different sound reproduction systems. In the analysis part, the diffuseness and direction of arrival of sound is estimated in a single location depending on time and frequency. In the synthesis part, microphone signals are first divided into non-diffuse and diffuse parts, and are then reproduced using different strategies. In this paper, the technology is reviewed, and implementations of DirAC for different applications are described. INTRODUCTION The coincident microphone approaches [1], such as first-order Ambisonics [2], can in theory utilize any loudspeaker setup. They are an attractive approach in spatial sound reproduction due to availability of goodquality microphones. Unfortunately, since the directional patterns of current high-quality microphones are only of zeroth or first order, the resulting loudspeaker signals of a multichannel setup are more coherent than desired. High coherence results in coloring and distortion of the spatial image, especially outside of the best listening position. The goal of the proposed Directional Audio Coding (DirAC) is to reproduce the spatial properties of sound recorded with a B-format microphone system as realistically as possible with different sound reproduction systems. DirAC performs some analysis of the sound field at frequency bands, and uses that information in reproduction, which circumvents some problems in traditional techniques. DirAC is based on the same principles and partly the same methods as the recently-proposed Spatial Impulse Response Rendering (SIRR) technique [3]. SIRR is a technique to reproduce room impulse responses for application in convolving reverberators. The first attempts to reproduce other sound than impulse responses with SIRR were presented in [4], which later lead to a system which is currently called Directional Audio Coding. Although DirAC was initially designed for high-quality applications, such as music reproduction etc., there are also some other applications where the technique can be used. As discussed in this paper, the technique can be used to transmit the spatial aspects of audio analyzed from a low-cost microphone grid as metadata, while the transmitted audio would only be a monophonic channel. This enables teleconferencing applications with spatial sound, as the transmitted data rate would not increase prominently from current solutions using also monophonic audio transmission. ASSUMPTIONS ABOUT PERCEPTION OF SPATIAL SOUND The design of the SIRR and DirAC techniques is based on four assumptions about the interaction between sound field properties and perceptual attributes that they produce [3]. The assumptions are repeated briefly here. The interested reader is referred to [5] for a discussion about their validity and for an introduction to the human mechanisms for spatial sound perception relevant in this case. For a more concise understanding of the perceptual mechanisms the reader is referred to, e.g. [6, 7]. 1 The assumptions are: 1 Direction of arrival of sound transforms into interaural time difference (ITD), interaural level difference (ILD), and monaural localization cues. 2 Diffuseness of sound transforms into interaural coherence cues. 3 Timbre depends on the monaural (time-dependent) spectrum together with ITD, ILD, and interaural coherence of sound. 4 The direction of arrival, diffuseness, and spectrum of sound measured in one position with the temporal and spectral resolution of human hearing determines the auditory spatial image the listener perceives. In SIRR, the number of sources used to measure the impulse response was always one, and the signal 1It has to be noted, that the term “diffuse” denotes only a quality of physical sound field in this paper. It is not used to quantify any perceptual aspect. 1 content of it was known to be an impulse. In DirAC, multiple sources are also allowed, and their content is not a priori known. However, the assumptions are not changed. It is still assumed that the listener cannot decode separate cues for multiple wavefronts from different directions within a critical band at one time instant. This is bolstered by psychoacoustic results, where it has been found, that two sinusoids with small spectral difference presented over spatially separated loudspeakers can not be localized correctly [8]. The listener typically localizes the sinusoids as a single fused event fluctuating inside the head between the ears. This makes us to assume that humans decode at one time only single cues per each critical band from ear canal signals. DirAC PROCESSING In this section, the processing parts of DirAC are presented. The general idea is first reviewed, after which different parts of processing are considered separately. Since there are multiple applications for DirAC, there are multiple variations for some of the parts. General idea of DirAC The DirAC design is based on the assumptions stated earlier in this paper, as was the SIRR design. In assumption 4 it was stated that the temporal and spectral resolution of the processing should mimic the temporal and spectral resolution the auditory system is using for spatial hearing. For this purpose, the microphone signals are divided into frequency bands, following the frequency decomposition of the inner ear as shown in Fig. 1. MICROPHONE CHANNELS IN TIME-FREQUENCY ANALYSIS MEASURED PROPERTIES REPRODUCTION AUDIO OUT Short-time Fourier analysis OR Filterbank DIRECTION and DIFFUSENESS ESTIMATION Energetic analysis elevation φ(t,f ) diffuseness ψ(t,f ) Point-like virtual sources Diffuse reproduction 1