Immersive Audio Signal Processing

This graduate-level text lays out the foundation of DSP for audio and the fundamentals of auditory perception, then goes on to discuss immersive audio rendering and synthesis, the digital equalization of room acoustics, and various DSP implementations. It covers a variety of topics and up-to-date results in immersive audio processing research: immersive audio synthesis and rendering, multichannel room equalization, audio selective signal cancellation, multirate signal processing for audio applications, surround sound processing, psychoacoustics and its incorporation in audio signal processing algorithms for solving various problems, and DSP implementations of audio processing algorithms on semiconductor devices.

Immersive Sound: The Art and Science of Binaural and Multi-Channel Audio provides a comprehensive guide to multi-channel sound. With contributions from leading recording engineers, researchers, and industry experts, Immersive Sound includes an in-depth description of the physics and psychoacoustics of spatial audio as well as practical applications. Chapters include the history of 3D sound, binaural reproduction over headphones and loudspeakers, stereo, surround sound, height channels, object-based audio, soundfield (ambisonics), wavefield synthesis, and multi-channel mixing techniques. Knowledge of the development, theory, and practice of spatial and multi-channel sound is essential to those advancing the research and applications in the rapidly evolving fields of 3D sound recording, augmented and virtual reality, gaming, film sound, music production, and post-production.

Audio Signal Processing for Next-Generation Multimedia Communication Systems presents cutting-edge digital signal processing theory and implementation techniques for problems including speech acquisition and enhancement using microphone arrays, new adaptive filtering algorithms, multichannel acoustic echo cancellation, sound source tracking and separation, audio coding, and realistic sound stage reproduction. This book's focus is almost exclusively on the processing, transmission, and presentation of audio and acoustic signals in multimedia communications for telecollaboration where immersive acoustics will play a great role in the near future.

Karlheinz Brandenburg and Mark Kahrs With the advent of multimedia, digital signal processing (DSP) of sound has emerged from the shadow of bandwidth limited speech processing. Today, the main appli cations of audio DSP are high quality audio coding and the digital generation and manipulation of music signals. They share common research topics including percep tual measurement techniques and analysis/synthesis methods. Smaller but nonetheless very important topics are hearing aids using signal processing technology and hardware architectures for digital signal processing of audio. In all these areas the last decade has seen a significant amount of application oriented research. The topics covered here coincide with the topics covered in the biannual work shop on “Applications of Signal Processing to Audio and Acoustics”. This event is sponsored by the IEEE Signal Processing Society (Technical Committee on Audio and Electroacoustics) and takes place at Mohonk Mountain House in New Paltz, New York. A short overview of each chapter will illustrate the wide variety of technical material presented in the chapters of this book. John Beerends: Perceptual Measurement Techniques. The advent of perceptual measurement techniques is a byproduct of the advent of digital coding for both speech and high quality audio signals. Traditional measurement schemes are bad estimates for the subjective quality after digital coding/decoding. Listening tests are subject to sta tistical uncertainties and the basic question of repeatability in a different environment.

When Speech and Audio Signal Processing published in 1999, it stood out from its competition in its breadth of coverage and its accessible, intutiont-based style. This book was aimed at individual students and engineers excited about the broad span of audio processing and curious to understand the available techniques. Since then, with the advent of the iPod in 2001, the field of digital audio and music has exploded, leading to a much greater interest in the technical aspects of audio processing. This Second Edition will update and revise the original book to augment it with new material describing both the enabling technologies of digital music distribution (most significantly the MP3) and a range of exciting new research areas in automatic music content processing (such as automatic transcription, music similarity, etc.) that have emerged in the past five years, driven by the digital music revolution. New chapter topics include: Psychoacoustic Audio Coding, describing MP3 and related audio coding schemes based on psychoacoustic masking of quantization noise Music Transcription, including automatically deriving notes, beats, and chords from music signals. Music Information Retrieval, primarily focusing on audio-based genre classification, artist/style identification, and similarity estimation. Audio Source Separation, including multi-microphone beamforming, blind source separation, and the perception-inspired techniques usually referred to as Computational Auditory Scene Analysis (CASA).

This open access book provides a concise explanation of the fundamentals and background of the surround sound recording and playback technology Ambisonics. It equips readers with the psychoacoustical, signal processing, acoustical, and mathematical knowledge needed to understand the inner workings of modern processing utilities, special equipment for recording, manipulation, and reproduction in the higher-order Ambisonic format. The book comes with various practical examples based on free software tools and open scientific data for reproducible research. The book’s introductory section offers a perspective on Ambisonics spanning from the origins of coincident recordings in the 1930s to the Ambisonic concepts of the 1970s, as well as classical ways of applying Ambisonics in first-order coincident sound scene recording and reproduction that have been practiced since the 1980s. As, from time to time, the underlying mathematics become quite involved, but should be comprehensive without sacrificing readability, the book includes an extensive mathematical appendix. The book offers readers a deeper understanding of Ambisonic technologies, and will especially benefit scientists, audio-system and audio-recording engineers. In the advanced sections of the book, fundamentals and modern techniques as higher-order Ambisonic decoding, 3D audio effects, and higher-order recording are explained. Those techniques are shown to be suitable to supply audience areas ranging from studio-sized to hundreds of listeners, or headphone-based playback, regardless whether it is live, interactive, or studio-produced 3D audio material.

Adaptive 3D Sound Systems focuses on creating multiple virtual sound sources in 3D reverberant spaces using adaptive filters. Adaptive algorithms are introduced and explained, including the multiple-error filtered-x algorithm and the adjoint LMS algorithm. The book covers the physical, psychoacoustical, and signal processing aspects of adaptive and non-adaptive 3D sound systems. Included is an introduction to spatial hearing, sound localization and reverberation, frequency selectivity of the human auditory system, the state of the art in HRTF-based 3D sound systems, binaural synthesis, and loudspeaker displays. The adaptive approach to HRTF-based 3D sound systems is examined in detail for the general case of creating multiple virtual sound sources at the ears of multiple listeners in a reverberant 3D space. The derived solution can be applied to other applications, such as cross-talk cancellation, loudspeakers and room equalization, concert hall simulation, and active sound control. Several solutions for the problem of moving listeners are introduced. Strategies for enlarging the zones of equalization around the listeners' ears, correct loudspeakers positioning, and using multiresolution filters are proposed. Fast multiresolution spectral analysis using non-uniform sampling is developed for implementation of multiresolution filters. The well-focused topics, along with implementation details for adaptive algorithms, make Adaptive 3D Sound Systems suitable for multimedia applications programmers, advanced level students, and researchers in audio and signal processing.