Abstract
This paper presents new results improving by a factor of 10 the accuracy of an Odd-DFT based frequency estimation algorithm. These results are shown to be robust to the influence of additive noise and compare favorably to other non-iterative frequency domain estimation algorithms. A perspective is given oil possible application areas, namely those involving real-time constraints.

Abstract
This paper presents several improvements that have been introduced to the design and operation of an adaptive 20-band room equalizer. The equalizer is implemented on a TMS320C6711 DSP platform and performs fast FIR filtering in the frequency domain. In order to reach fast adaptation to time-varying acoustic conditions, several adaptation rules operating in the frequency domain have been evaluated and the impact of a frequency-varying stepsize parameter on the convergence rate has been studied. These results will be presented along with ideas and plans for future developments.

Abstract
In the application of conventional audio compression algorithms to low bit rate audio coding one is faced with the unsatisfactory tradeoff between coarser quantization and audio bandwidth reduction. Frequency Extension has therefore emerged as an important tool for the satisfactory performance of low bit rate audio codecs. In this paper we describe one of a newer class of Frequency Extension techniques which are applied directly to the high frequency resolution representation of the signal (e.g., MDCT). This particular technique is based on a Fractal Self-Similarity Model (FSSM) for the short-term frequency representation of the signal. The FSSM model, which may include multiple dilation and translation terms, has been found to be effective for a wide variety of speech and music signals and provides a compact description for long term correlation that may exist in frequency domain. The high frequency resolution of MDCT aids in accurate parameter estimation for the model, which in turn has shown promise as a Frequency Extension tool that offers a detailed and natural sounding quality at low bit rates. Structure of the FSSM model, issues related to parameter estimation, and its application to audio coding for bit rates of 8-48 kbps is discussed. Audio demos are available at http://www.atc-labs.com/fssm.

Abstract
This paper presents a new method to the adaptive cancellation of acoustic feedbacks. The method uses high resolution frequency analysis and high-Q notch filters so as to accurately detect feedbacks and cancel them without disturbing noticeably the main audio spectrum. The method will be described, its implementation on a TMS320C6711 DSP platform for real time operation will be explained, and results for the adaptive cancellation of two simultaneous acoustic feedbacks will be presented.

Abstract
Recent advances in perceptual audio coding are strongly based on the concept of bandwidth extension. Most techniques implementing bandwidth extension require an analysis/synthesis filter bank in addition to that used by the associated perceptual audio coder, which increases the overall system complexity and coding delay, and makes difficult the correct alignment between the operation of the audio coder and the operation of the bandwidth extension technique. We present a new Accurate Spectral Replacement (ASR) technique that is based on a suitable decomposition of the MDCT filter bank, and that implements synthesis of sinusoidal components with an accuracy much higher than the natural frequency resolution of the filter bank. The ASR technique is described, its performance is assessed with both synthetic and natural audio signals, and its main areas of application are addressed. Audio demos are available at http://www.atc-labs.com/asr/.

Abstract
High-quality audio bit-rate reduction systems are widely used in many application areas involving audio broadcast, streaming and download services. With the advent of 3G mobile and wireless communication networks, there is a clear opportunity for new multimedia services, notably those relying on two-way high- quality audio communication. In t his paper we describe a new source/perceptual audio coder that features low-delay, intrinsic error robustness and high subjective audio quality at competitive compression ratios. The structure of the audio coder is described and an emphasis is given on its innovative approaches to semantic signal segmentation and decomposition, independent coding of sinusoidal and noise components, and bandwidth extension using Accurate Spectral Replacement. A few test results are presented that illustrate the operation and performance of the new coder.