An interactive audio service is a new conceptual audio service that provides the users with opportunities for a variety of experiences on the alternative and advanced audio services. In the interactive audio service, users can freely control various audio objects to make their own audio sounds. A spatial audio object coding (SAOC) is a useful technology that can support most parts of the interactive audio service with a relatively low bit-rate, but is very poor to perfect gain control of a certain audio object, i.e., the target audio object. In this paper, the SAOC with a two-step coding structure is proposed to efficiently handle the target audio object as well as the normal audio objects. A transform coded excitation (TCX) based residual coding scheme is presented in the context of the sound quality enhancement. From experimental results, it can be noted that the various audio objects can be successfully handled with respect to the bit-rate and the sound quality by using the proposed two-step coding structure SAOC.

An unsupervised method is proposed aiming at extracting rhythmic sources from commercial polyphonic music whose number of channels is limited to one. Commercial music signals are not usually provided with more than two channels while they often contain multiple instruments including singing voice. Therefore, instead of using conventional ways, such as modeling mixing environments or statistical characteristics, we should introduce other source-specific characteristics for separating or extracting the sources. In this paper, we concentrate on extracting rhythmic sources from the mixture with the other harmonic sources. An extension of nonnegative matrix factorization (NMF) is used to analyze multiple relationships between spectral and temporal properties in the given input matrices. Moreover, temporal repeatability of the rhythmic sound sources is implicated as common rhythmic property among segments of an input mixture signal. The proposed method shows acceptable, but not superior separation quality to the referred drum source separation systems. However, it has better applicability due to its blind manner in separation.

We address a problem of separating drums from polyphonic music containing various pitched instruments as well as drums. Nonnegative matrix factorization (NMF) was successfully applied to spectrograms of music to learn basis vectors, followed by support vector machine (SVM) to classify basis vectors into ones associated with drums (rhythmic source) only and pitched instruments (harmonic sources). Basis vectors associated with pitched instruments are used to reconstruct drum-eliminated music. However, it is cumbersome to construct a training set for pitched instruments since various instruments are involved. In this paper, we propose a method which only incorporates prior knowledge on drums, not requiring such training sets of pitched instruments. To this end, we present nonnegative matrix partial co-factorization (NMPCF) where the target matrix (spectrograms of music) and drum-only-matrix (collected from various drums a priori) are simultaneously decomposed, sharing some factor matrix partially, to force some portion of basis vectors to be associated with drums only. We develop a simple multiplicative algorithm for NMPCF and show its usefulness empirically, with numerical experiments on real-world music signals.

Binaural cue coding (BCC) has been introduced for compact representation of multi-channel audio. It exploits binaural cue parameters for capturing the spatial image of multi-channel audio. Recently, it has been standardized within MPEG as the name of "MPEG Surround." In this paper, we propose a sound source location cue coding (SSLCC) system for compressing multi-channel audio to be suitable at the narrow bandwidth transmission environment. To improve the compression ability of the conventional BCC, the SSLCC system utilizes the virtual source location information (VSLI) as a spatial cue parameter instead of the inter-channel level difference (ICLD) of the BCC system. Also the SSLCC system adopts enhanced pre/post processing algorithms to improve perceptual sound quality. Objective and subjective assessment results show that the proposed SSLCC system reveals better performance than the conventional BCC system.

Detecting and classifying edge components in images find a wide range of applications from perceptually improved coding schemes to content-based searches. In this paper, a fast and systematic scheme is proposed to classify the edge orientation of each block in discrete cosine transform (DCT)-compressed images. Derived from a simple, but acceptable, pixel-domain algorithm, the proposed scheme performs directly on DCT coefficient domain and, thus, saves the arithmetic operations by a considerable amount. The performance of the proposed method is validated by the experiments against various natural images.

Recently, FD (frame dropping)-CD (coefficient dropping) transcoding has been considered, mainly due to the low computational complexity and simple implementation. However, CD errors in the FD-CD transcoding scheme tend to be propagated and they have a significant effect on the qualities of decoded images. We derive the error characteristics incurred by CD operations and propose an effective estimation model that adaptively describes well the characteristics of propagation errors in the compressed domain. Furthermore, we apply the proposed model to distortion control achieving nearly constant distortion allocation among frames. Simulations show that the proposed model is quite accurate in estimating the overall distortions and is effectively applied to distortion control over a range of sequences with varying scene types.

In this paper, we describe the real-time implementation effort of MPEG-4 audio HVXC (Harmonic Vector eXcitation Coding) encoder algorithm for very low bitrates, which has target applications from mobile communications to Internet telephony, on current high performance floating point DSP. Using the C-language and assembly-language level optimization for time-critical functional codes, utilizing the internal program memory of the DSP as the program cache, and further utilizing the internal data memory operation and DMA functionality we could get a goal of realtime operation of HVXC encoder both at 2 kbit/s and at 4 kbit/s.