摘要:
A method for applying a computation utilizing a processor (112) capable of accessing an on-chip memory (114) and data from an off-chip source (116), the method comprising the iterative steps of retrieving at the on-chip memory (114) successive frames of input data from the off-chip source (116); computing from a current input frame of data (30) available in the on-chip memory current result elements for completing a current output frame of results (33, 34, 35), and pre-computing from the current frame of input data future result elements for contributing to at least one future output frame of results.
摘要:
A method to process audio signals, an apparatus accepting audio signals, a carrier medium that carried instructions for a processor to implement the method to process audio signals, and a carrier medium carrying filter data to implement a filter of audio signals. The method includes filtering a pair of audio input signals by a process that produces a pair of output signals corresponding to the results of: filtering each of the input signals with a HRTF filter pair, and adding the HRTF filtered signals. The HRTF filter pair is such that a listener listening to the pair of output signals through headphones experiences sounds from a pair of desired virtual speaker locations. Furthermore, the filtering is such that, in the case that the pair of audio input signals includes a panned signal component, the listener listening to the pair of output signals through headphones is provided with the sensation that the panned signal component emanates from a virtual sound source at a center location between the virtual speaker locations.
摘要:
Sound source localization apparatuses and methods are described. A frame amplitude difference vector is calculated based on short time frame data acquired through an array of microphones. The frame amplitude difference vector reflects differences between amplitudes captured by microphones of the array during recording the short time frame data. Similarity between the frame amplitude difference vector and each of a plurality of reference frame amplitude difference vectors is evaluated. Each of the plurality of reference frame amplitude difference vectors reflects differences between amplitudes captured by microphones of the array during recording sound from one of a plurality of candidate locations. A desired location of sound source is estimated based at least on the candidate locations and associated similarity. The sound source localization can be performed based at least on amplitude difference.
摘要:
An improved decorrelator is disclosed that processes an input audio signal in two separate paths. In one path, a banded phase-flip filter is applied to lower frequencies of the input audio signal. In a second path, a frequency-dependent delay is applied to higher frequencies of the input audio signal. Signals from the two paths are combined to obtain an output signal that is psychoacoustically decorrelated with the input audio signal. The decorrelated signal can be mixed with the input audio signal without generating audible artifacts.
摘要:
A method of producing an approximation sequence to a series of sample values, the method comprising the steps of (a) determining a first set having candidate partial sequences as members, each member comprising a plurality of elements; (b) selecting the first n elements of one of the members of the first set as a next output element for said approximation sequence; n a positive integer; (c) forming a second set having descendent candidate partial sequences as members from said first set; (d) applying a fitness filtering process to said second set to rank its members according to fitness for representing at least a corresponding portion of the series of input samples; (e) selecting at least some of the members of the second set to form a third set; and repeating steps (a) to (e) so as to produce said approximation sequence, wherein the third set of step (e) functions as the first set of the subsequent step (a).
摘要:
An improved method and apparatus for filtering an electronic environment with relatively high accuracy and efficiency and relatively short flow-through delay ("latency") is disclosed. Embodiments of the invention may be applied to digital filters implemented in software, hardware or a combination of both for applications such as audio filtering or electronic modelling of acoustic system characteristics. The method disclosed is broadly applicable in the field of signal processing and may be used to advantage, for example, in adaptive filtering; audio reverberation processing; adaptive echo cancellation; spatial processing; virtual reality audio; correlation, radar; radar pulse compression; deconvolution; seismic analysis; telecommunications; pattern recognition; robotics; 3D acoustic modelling; audio post production (including auralization and auto reverberant matching); audio equalization; compression; sonar; ultrasonics; secure communication systems; digital audio broadcast, acoustic analysis, surveillance; noise cancellation; and echo cancellation.
摘要:
Sound source localization apparatuses and methods are described. A frame amplitude difference vector is calculated based on short time frame data acquired through an array of microphones. The frame amplitude difference vector reflects differences between amplitudes captured by microphones of the array during recording the short time frame data. Similarity between the frame amplitude difference vector and each of a plurality of reference frame amplitude difference vectors is evaluated. Each of the plurality of reference frame amplitude difference vectors reflects differences between amplitudes captured by microphones of the array during recording sound from one of a plurality of candidate locations. A desired location of sound source is estimated based at least on the candidate locations and associated similarity. The sound source localization can be performed based at least on amplitude difference.
摘要:
In some embodiments, a method for generating a matrix-encoded two-channel audio signal in response to a horizontal B-format signal by performing a mixing operation. In other embodiments, a method for generating a matrix-encoded two-channel audio signal, including steps of generating microphone output signals (by capturing sound with a microphone array), and performing a mixing operation on the microphone output signals, where the mixing operation is equivalent to generating a horizontal B-format signal in response to the microphone output signals, and generating the matrix-encoded two-channel audio signal in response to the horizontal B-format signal. The microphone array is typically a small array of cardiod microphones (e.g., an array consisting of three cardiod microphones). Other aspects include systems (e.g., encoders) programmed or otherwise configured to perform any embodiment of the method for generating a matrix-encoded two-channel audio signal.
摘要:
The loudness of an audio signal is adjusted while reducing changes in its perceived spectral balance, using a dynamically-controllable filter having a high-frequency response characteristic and a low-frequency response characteristic, controlled by dynamically-changing information on the desired gain in each of a plurality of frequency bands of the audio signal.
摘要:
Frequency-domain techniques are used for adaptive equalization that is responsive to spectral magnitude characteristics but not sensitive to phase characteristics of system response. Signal correlation may be used to improve adaptation accuracy when significant levels of ambient sounds are present. A preferred filter implementation uses convolution-based block transforms and cross-fade windows.