AES San Francisco 2008
Poster Session P24
P24 - Audio Digital Signal Processing and Effects—Part 1
Sunday, October 5, 9:00 am — 10:30 am
P24-1 Simple Arbitrary IIRs—Richard Lee, Pandit Littoral - Cooktown, Queensland, Australia
This is a method of fitting IIRs (Infinite Impulse Response filters) to an arbitrary frequency response simple enough to incorporate in intelligent AV receivers. Short IIR filters are useful where computational power is limited and at low frequencies where FIRs have poor performance. Loudspeaker and microphone frequency response defects are often better matched to IIRs. Some caveats for digital EQ design are discussed. The emphasis is on loudspeakers and microphones
Convention Paper 7635 (Purchase now)
P24-2 Analysis of Design Parameters for Crosstalk Cancellation Filters Applied to Different Loudspeaker Configurations—Yesenia Lacouture Parodi, Aalborg University - Aalborg, Denmark
Several approaches to render binaural signals through loudspeakers have been proposed in past decades. Some studies have focused on the optimum loudspeaker arrangement while others have proposed more efficient filters. However, to our knowledge, the identification of optimal parameters for crosstalk cancellation filters applied to different loudspeaker configurations has not yet been addressed systematically. In this paper we document a study of three different inversion techniques applied to several loudspeaker arrangements. Least square approximations in frequency and time domain are evaluated along with a crosstalk canceller-based on minimum-phase approximation. The three methods are simulated in two-channel configuration and the least square approaches in four-channel configurations. Different span angles and elevations are evaluated for each case. In order to obtain optimum parameter, we varied the bandwidth, filter length, and regularization constant for each loudspeaker position and each method. We present a description of the simulations carried out and the optimum regularization values, expected channel separation, and performance error obtained for each configuration.
Convention Paper 7636 (Purchase now)
P24-3 A Hybrid Time and Frequency Domain Audio Pitch Shifting Algorithm—Nicolas Juillerat, University of Fribourg - Fribourg, Switzerland; Stefan Müller Arisona, University of Santa Barbara - Santa Barbara, CA, USA; Simon Schubiger-Banz, Computer Systems Institute, ETH Zürich - Zürich, Switzerland
This paper presents an abstract algorithm that performs audio pitch shifting as a combination of a signal analysis, a filter bank, and frequency shifting operations. Then, it is shown that two previously proposed pitch shifting algorithms are actually concrete implementations of the presented abstract algorithm. One of them is implemented in the frequency domain whereas the other is implemented in the time domain. Based on an analysis and comparison of the properties of these two implementations (quality, artifacts, assumptions on the signal), we propose a new hybrid implementation working partially in the frequency domain and partially in the time domain, and achieving superior quality by taking the best from each of the two existing implementations.
Convention Paper 7637 (Purchase now)
P24-4 A Colored Noise Suppressor Using Lattice Filter with Correlation Controlled Algorithm—Arata Kawamura, Youji Iiguni, Osaka University - Toyonaka, Osaka, Japan
A noise suppression technique is necessary in a wide range of applications including mobile communication and speech recognition systems. We have previously proposed a noise suppressor using a lattice filter that can cancel a white noise from an observed signal. Unfortunately, many practical noises are not white, and hence the conventional noise suppressor is not available for the practical noises. In this paper we propose a new adaptive algorithm used for the lattice filter to suppress a colored noise. The proposed algorithm can be directly derived from the conventional time recursive algorithm. To extract a speech from a speech mixed with colored noise, the lattice filter with the proposed algorithm gives a noise replica whose auto-correlation is close to the noise’s one. Subtracting the noise replica from the observed noisy speech, we can obtain an extracted speech. Simulation results showed that the proposed noise suppressor can extract a speech from a speech mixed with a tunnel noise, which is a colored noise recorded in a practical environment.
Convention Paper 7638 (Purchase now)
P24-5 Accurate IIR Equalization to an Arbitrary Frequency Response, with Low Delay and Low Noise Real-Time Adjustment—Peter Eastty, Oxford Digital Limited - Stonesfield, Oxfordshire, UK
A new form of equalizer has been developed that combines minimum phase, low delay, IIR signal processing with low noise, real-time adjustment of coefficients to accurately deliver an arbitrary frequency response as entered from a graphical user interface. The use of a join-the-dots type graphical user interface combined with cubic or similar splines is a common method of entering curved lines into 2-D drawing programs. The equalizer described in this paper combines a similar type of user interface with low-delay, minimum phase, IIR audio DSP. Key attributes also include real-time, nearly noiseless adjustment of the DSP coefficients in response to user input. All necessary information for the construction of these filters is included.
Convention Paper 7639 (Purchase now)
P24-6 A Method of Capacity Increase for Time-Domain Audio Watermarking Based on Low-Frequency Amplitude Modification—Harumi Murata, Akio Ogihara, Motoi Iwata, Akira Shiozaki, Osaka Prefecture University - Osaka, Japan
The objective of this work is to increase the capacity of watermark information in “the audio watermarking method based on amplitude modification,” which has been proposed by W. N. Lie as a prevention technique against copyright infringement. In this conventional method, the capacity of watermark information is not enough, and it is desirable that the capacity of watermark information is increased. In this paper we increase the capacity of watermark information by embedding multiple watermarks in the different levels of audio data independently. The proposed method has many data-channels for embedding, and hence it is possible to embed multiple watermarks by selecting the proper data-channel according to required data capacity or recovery rate.
Convention Paper 7640 (Purchase now)
P24-7 Constrained-Optimized Sound Beamforming of Loudspeaker-Array System—Myung Song, Soonho Baek, Yonsei University - Seoul, Korea; Seok-Pil Lee, Korea Electronics Technology Institute - Seongnam, Korea; Hong-Goo Kang, Yonsei University - Seoul, Korea
This paper proposes a novel loudspeaker-array system to form relatively high sound pressure toward the desired location. The proposed algorithm adopts a constrained-optimization technique such that the array response to the desired response is maintained over mainlobe width while minimizing its sidelobe level. At first the characteristic of sound propagation in reverberant environment is analyzed by off-line computer simulation. Then, the performance of the implemented loudspeaker-array system is evaluated by measuring sound pressure distribution in a real test room. The results show that the proposed sound beamforming algorithm forms more concentrative sound beam to the desired location than conventional algorithms even in a reverberation environment.
Convention Paper 7641 (Purchase now)