Audio signal processing

About: Audio signal processing is a research topic. Over the lifetime, 21463 publications have been published within this topic receiving 319597 citations. The topic is also known as: audio processing & Acoustic signal processing.

An Introduction to Audio Content Analysis: Applications in Signal Processing and Music Informatics

Abstract: With the proliferation of digital audio distribution over digital media, audio content analysis is fast becoming a requirement for designers of intelligent signal-adaptive audio processing systems. Written by a well-known expert in the field, this book provides quick access to different analysis algorithms and allows comparison between different approaches to the same task, making it useful for newcomers to audio signal processing and industry experts alike. A review of relevant fundamentals in audio signal processing, psychoacoustics, and music theory, as well as downloadable MATLAB files are also included. Please visit the companion website: www.AudioContentAnalysis.org

Computer vision for music identification

Abstract: We describe how certain tasks in the audio domain can be effectively addressed using computer vision approaches. This paper focuses on the problem of music identification, where the goal is to reliably identify a song given a few seconds of noisy audio. Our approach treats the spectrogram of each music clip as a 2D image and transforms music identification into a corrupted sub-image retrieval problem. By employing pairwise boosting on a large set of Viola-Jones features, our system learns compact, discriminative, local descriptors that are amenable to efficient indexing. During the query phase, we retrieve the set of song snippets that locally match the noisy sample and employ geometric verification in conjunction with an EM-based "occlusion" model to identify the song that is most consistent with the observed signal. We have implemented our algorithm in a practical system that can quickly and accurately recognize music from short audio samples in the presence of distortions such as poor recording quality and significant ambient noise. Our experiments demonstrate that this approach significantly outperforms the current state-of-the-art in content-based music identification.

Multipitch Analysis of Polyphonic Music and Speech Signals Using an Auditory Model

Abstract: A method is described for estimating the fundamental frequencies of several concurrent sounds in polyphonic music and multiple-speaker speech signals. The method consists of a computational model of the human auditory periphery, followed by a periodicity analysis mechanism where fundamental frequencies are iteratively detected and canceled from the mixture signal. The auditory model needs to be computed only once, and a computationally efficient strategy is proposed for implementing it. Simulation experiments were made using mixtures of musical sounds and mixed speech utterances. The proposed method outperformed two reference methods in the evaluations and showed a high level of robustness in processing signals where important parts of the audible spectrum were deleted to simulate bandlimited interference. Different system configurations were studied to identify the conditions where pitch analysis using an auditory model is advantageous over conventional time or frequency domain approaches.

Multi-channel audio decoder

Abstract: A subband audio coder employs perfect/non-perfect reconstruction filters, predictive/non-predictive subband encoding, transient analysis, and psycho-acoustic/minimum mean-square-error (mmse) bit allocation over time, frequency and the multiple audio channels to encode/decode a data stream to generate high fidelity reconstructed audio. The audio coder windows the multi-channel audio signal such that the frame size, i.e. number of bytes, is constrained to lie in a desired range, and formats the encoded data so that the individual subframes can be played back as they are received thereby reducing latency. Furthermore, the audio coder processes the baseband portion (0-24 kHz) of the audio bandwidth for sampling frequencies of 48 kHz and higher with the same encoding/decoding algorithm so that audio coder architecture is future compatible.

Method of providing identification, other digital data and multiple audio tracks in video systems

Abstract: A method and apparatus for imbedding digital data and multiple audio (analog) track information in a video signal in a manner compatible with ordinary broadcast TV and transparent to a conventional television receiver is disclosed. The method and apparatus may provide multiple independent audio tracks and a substantial level of interaction with a viewer utilizing special reception equipment, which equipment may be utilized to provide a fully interactive system from signal sources not required to be compatible with convention receivers. The method and apparatus includes the ability of imbedding both analog and digital signals in a video signal, and while advantageous for interactive TV systems, may be used for a multitude of purposes in various video systems.