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.

Multichannel nonnegative tensor factorization with structured constraints for user-guided audio source separation

Abstract: Separating multiple tracks from professionally produced music recordings (PPMRs) is still a challenging problem. We address this task with a user-guided approach in which the separation system is provided segmental information indicating the time activations of the particular instruments to separate. This information may typically be retrieved from manual annotation. We use a so-called multichannel nonnegative tensor factorization (NTF) model, in which the original sources are observed through a multichannel convolutive mixture and in which the source power spectrograms are jointly modeled by a 3-valence (time/frequency/source) tensor. Our user-guided separation method produced competitive results at the 2010 Signal Separation Evaluation Campaign, with sufficient quality for real-world music editing applications.

Signaling audio rendering information in a bitstream

Abstract: In general, techniques are described for specifying audio rendering information in a bitstream. A device configured to generate the bitstream may perform various aspects of the techniques. The bitstream generation device may comprise one or more processors configured to specify audio rendering information that includes a signal value identifying an audio renderer used when generating the multi-channel audio content. A device configured to render multi-channel audio content from a bitstream may also perform various aspects of the techniques. The rendering device may comprise one or more processors configured to determine audio rendering information that includes a signal value identifying an audio renderer used when generating the multi-channel audio content, and render a plurality of speaker feeds based on the audio rendering information.

Analog memory for storing digital information

Abstract: The present invention is directed to an analog memory for storing digital information in analog signal form. Typically, digital information is stored in digital signal form, where each digital bit is stored in a separate digital memory cell. In accordance with the present invention, an analog memory such as a charge transfer device (CTD), bubble memory, or magnetostrictive memory is used to store analog signals. Each analog signal is representative of a plurality of digital bits, thereby providing storage for a plurality of digital bits in each analog memory cell. Use of such an analog memory in combination with a digital system facilitates a hybrid memory, where digital information is stored in analog signal form. In one embodiment, a digital to analog converter is used to convert digital information from a digital processor to analog signal form for storage in an analog memory and an analog to digital converter is used to convert analog signals stored in the analog memory to digital signal form for processing with the digital processor. In another embodiment, an analog read only memory is used to store a program for a stored program digital computer in analog signal form. Storage of digital information in analog signal form increases the efficiency of storage because a plurality of digital bits can be stored in each memory cell. An embodiment having analog error compensation utilizes a reference signal for adaptive compensation of errors. Various systems using such memories are disclosed including signal processors, stored program computers, reverbation systems, and others.

Audio Fingerprinting: Combining Computer Vision & Data Stream Processing

Abstract: In this paper, we present waveprint, a novel system for audio identification. Waveprint uses a combination of computer-vision techniques and large-scale-data-stream processing algorithms to create compact fingerprints of audio data that can be efficiently matched. The resulting system has excellent identification capabilities for small snippets of audio that have been degraded in a variety of manners, including competing noise, poor recording quality, and cell-phone playback. We measure the tradeoffs between performance, memory usage, and computation through extensive experimentation. The system is more efficient in terms of memory usage and computation, while being more accurate, when compared with previous state of the art systems.

Unequal error protection for perceptual audio coders

Abstract: An audio information bit stream including audio control bits and audio data bits is processed for transmission in a communication system. The audio data bits are first separated into n classes based on error sensitivity, that is, the impact of errors in particular audio data bits on perceived quality of an audio signal reconstructed from the transmission. Each of the n different classes of audio data bits is then provided with a corresponding one of n different levels of error protection, where n is greater than or equal to two. The invention thereby matches error protection for the audio data bits to source and channel error sensitivity. The audio control bits may be transmitted independently of the audio data bits, using an additional level of error protection higher than that used for any of the n classes of the audio data bits. Alternatively, the control bits may be combined with one of the n classes of audio data bits and provided with the highest of the n levels of error protection. Further protection may be provided for the control bits by repeating at least a portion of the control bits from a current packet of the audio information bit stream in a subsequent packet of the audio information bit stream. Moreover, the classification of audio data bits into n different classes can be implemented on a fixed packet-by-packet basis, or in a more flexible, adaptive implementation in which different multipacket error protection profiles are used for different multipacket segments of a source-coded audio signal.