Abstract: A system and method for generating virtual microphone signals having a particular number and configuration for channel playback from an intermediate set of signals that were recorded in an initial format that is different from the channel playback format. In one embodiment, an initial set of intermediate are Bark-banded such that each intermediate signal may lead to a corresponding power spectral density (PSD) signal representative of the initial intermediate signal. Further, one may generate cross-correlations signals for each pair of intermediate signals. Next, from the PSDs and cross correlations, one may more efficiently calculate corresponding channel signals to be used for playback on respective channel speakers. Thus, the PSDs of each channel signal may be generated at chosen angles (as well as other design factors). Further, each channel signal may also be further modified with a corresponding cancellation signal that further enhances the resultant signal in each channel.

Abstract: Perceptual audio coder refers to audio compression schemes that exploit the properties of human auditory perception. The coder allocates the quantization noise below the masking threshold such that even with the bit rate limitation, the noise is imperceptible to the ear. These distortion and bit rate requirement makes the bit allocation-quantization process a considerable computational effort. One method includes incrementally adjusting a global gain according to a gradient. The gradient could be adjusted each time the number of bits used to represent a quantized value is counted. Another method includes limiting a rate controlling parameter to a predetermined number of loops. The method could also include deriving a global gain to ensure exit from the loop. Accordingly, embodiments of the present disclosure provide a fast and efficient method to derive the rate controlling parameter and can be applied to generic perceptual audio encoders where low computational complexity is required.

Abstract: The present disclosure provides a digital audio signal processing system that comprises a set of delay lines, allpass and lowpass filters to achieve the reverberation effect. The present disclosure further provides a method for generating and controlling digital reverberations for audio signals. The reverberation generated will have an increasing echo density in the time domain and a faster decay of high frequency signals than low frequency signals. The controlling mechanism of reverberation generation is realized through the extraction of the real environment characteristics.

Abstract: The invention provides for the encoding of surround sound produced by any coincident microphone techniques with coincident-to-virtual microphone signal matrixing. An encoding scheme provides significantly lower computational demand, by deriving the spatial parameters and output downmixes from the coincident microphone array signals and the coincident-to-surround channel-coefficients matrix, instead of the multi-channel signals.

Abstract: The quality of music output from audio systems is improved by simulating the effect of low frequency signals in the human ear. This thus allows listeners to perceive the lower frequency signals, even though the speakers may be incapable of providing such low frequency outputs. Method and systems provided for processing enhancing bass effect in audio signals. Said method and systems result in the bass enhancement being computationally less intensive. The bass effect enhancement techniques described are based on the response of sine and cosine transfer functions and on the directional independence of low frequency components. The human ear is unable to resolve directions from low frequency components. The bass effect enhancement technique alternatively is based on response of an exponential transfer function.

Abstract: A method for stereo audio perceptual encoding of an input signal includes masking threshold estimation and bit allocation. The masking threshold estimation and bit allocation are performed once every two encoding processes. Another method for stereo audio perceptual encoding of an input signal includes performing a time-to-frequency transformation, performing a quantization, performing a bitstream formatting to produce an output stream, and performing a psychoacoustics analysis. The psychoacoustics analysis includes masking threshold estimation on a first of every two successive frames of the input signal.

Abstract: In order to reproduce audio signals which have been compressed or encoded for storage or transmission using, for example, MPEG audio encoding, a synthesis sub-band filter is employed which performs an inverse modified discrete cosine transform. The computational cost of the IMDCT implementation is reduced by pre-calculating arrays of sum and difference data. The arrays of sum and difference data are then used in two separate transform calculations, the results of which can be used in the generation of pulse code modulation audio data.

Abstract: The quality of music output from audio systems is improved by simulating the effect of low frequency signals in the human ear. This thus allows listeners to perceive the lower frequency signals, even though the speakers may be incapable of providing such low frequency outputs. A method is provided for processing enhancing bass effect in audio signals. The method also results in the bass enhancement being computationally less intensive. The bass effect enhancement techniques are based on the response of sine and cosine transfer functions and on the directional independence of low frequency components. The human ear is unable to resolve directions from low frequency components. The bass effect enhancement technique alternatively is based on the response of an exponential transfer function.

Abstract: An apparatus for adaptively suppressing noise in an input signal frequency spectrum derived from overlapping input frames is provided. The system includes a psychoacoustic power computation module configured to compute a noisy signal power in psychoacoustic bands, a voice activity scoring module configured to compute a probabilistic score for a presence of a speech, and a noise estimation module configured to estimate a noise power in the psychoacoustic bands based on information of past frames, the probabilistic score, and the computed noisy signal power. The system also includes a gain computation module configured to compute a gain for each frequency, based on a probabilistic heuristic, the probabilistic score and the information on the past frames, and a gain post-processing module configured to perform a gain time smoothing, a gain frequency smoothing, and a gain regulation for the computed gain.

Abstract: Methods and apparatuses for detection and reduction of wind noise in audio devices are disclosed. In an embodiment, a method includes acquiring and transforming the audio signals. Correlations from the transformed audio signals are computed. A cross correlation index is compared to a predetermined value to determine if a wind noise spectral content is present. In another embodiment, an apparatus includes an audio processing unit to receive non-decomposed audio signals, and an audio decomposition unit to receive the non-decomposed audio signals and to generate decomposed audio signals. A wind noise spectrum estimation unit receives non-decomposed audio signals and decomposed audio signals and identifies wind noise spectral components in at least one of the non-decomposed and decomposed audio signals. A wind noise spectrum reduction unit receives the wind noise spectral components and removes the wind noise spectral components from at least one of the non-decomposed and the decomposed audio signals.

Abstract: A method and audio device are presented that preserve mono energy during downmixing of a hybrid coding process of an audio signal. The method includes calculating a stereo scaling factor in a group level that is definable within a stereo band. The method may also include updating the stereo scaling factor using an update rate and synchronizing the update rate of a spatial parameter during a fast changing transient portion of the signal. A number of groups in a first stereo band may be greater than a number of groups in a second stereo band, and the first stereo band may be a lower frequency band than the second band or may be perceptually more important than the second band.

Abstract: The present invention provides an audio streaming system and method for transmitting audio signals with high quality. The advantages of the present invention include easy implementation, computational efficiency, and provision of better audio quality. More particularly, the present invention provides a Multi-band Time Expansion algorithm for lost packet concealment. The Multi-band Time Expansion algorithm detects the number of continuously lost packets in an audio input signal and the correctly received packets on either side of the lost packets. Then the Multi-band Time Expansion algorithm time-expands the correctly received packets that may be from either one side or both sides of the lost packets, wherein the correctly received packets are stretched to cover the length of the lost packets. Finally the Multi-band Time Expansion algorithm overlap-adds the stretched packets so that the lost packets are concealed.

Abstract: An apparatus, method, and computer program are capable of receiving and cross-correlating a first audio signal and a second audio signal. This produces a cross-correlated signal, which is used to identify a plurality of parameters associated with at least one of the first and second audio signals. The parameters are used to generate an indicator identifying an extent to which the first and second audio signals match.

Abstract: A method includes splitting and filtering a left input signal and a right input signal to produce a plurality of frequency sub-bands. Each of the frequency sub-bands includes a left sub-band signal and a right sub-band signal. The method also includes processing the left and right sub-band signals associated with each of the frequency sub-bands into a plurality of sub-band channel signals. The plurality of sub-band channel signals includes at least three sub-band channel signals. In addition, the method includes summing corresponding ones of the sub-band channel signals for reproduction in a corresponding channel of a plurality of channels. The plurality of sub-band channel signals may include a left sub-band channel signal, a right sub-band channel signal, a center sub-band channel signal, a left surround sub-band channel signal, and a right surround sub-band channel signal.

Abstract: A system and method of reusing information in low power scalable hybrid audio encoders. The system and method provides a transform coder and parameterization of high frequency spectrum (SBR).

Abstract: An audio processor, apparatus, and method use physical speakers to emulate one or more additional speakers. The physical speakers produce sounds that, from a listener's perspective, appear to come from at least one direction where a physical speaker is not present. Any number of additional speakers can be virtualized, such as three or five speakers that allow two speakers to emulate a 5.1 audio system.

Abstract: The perception of 3D sound positioning can be achieved using a 2D arrangement of speakers positioned around the listener. The disclosed techniques can enable listeners to perceive sounds as coming from above and/or below them, without the need for positioning speakers above and/or below the listener. In some embodiments, elevation information can be included in the X and Y horizontal components of the 2D ambisonics encoding. The X and Y components can be decoded using 2D ambisonics decoding. Suitable filtering may be performed on the decoded sound information to enhance the listener's perception of the elevation information encoded in the X and Y components.

Abstract: The quality of music output from audio systems is improved by simulating the effect of low frequency signals in the human ear. This thus allows listeners to perceive the lower frequency signals, even though the speakers may be incapable of providing such low frequency outputs. Method and systems provided for processing enhancing bass effect in audio signals. Said method and systems result in the bass enhancement being computationally less intensive. The bass effect enhancement techniques described are based on the response of sine and cosine transfer functions and on the directional independence of low frequency components. The human ear is unable to resolve directions from low frequency components. The bass effect enhancement technique alternatively is based on response of an exponential transfer function.

Abstract: A time-domain method of adaptively levelling the loudness of a digital audio signal is proposed. It selects a proper frequency weighting curve to relate the volume level to the human auditory system. The audio signal is segmented into frames of a suitable duration for content analysis. Each frame is classified to one of several predefined states and events of perceptual interest is detected. Four quantities are updated each frame according to the classified state and detected event to keep track of the signal. One quantity measures the long-term loudness and is the main criterion for state classification of a frame. The second quantity is the short-term loudness that is mainly used for deriving the target gain. The third quantity measures the low-level loudness when the signal is deemed to not contain important content, giving a reasonable estimate of noise floor. A fourth quantity measures the peak loudness level that is used to simulate the temporal masking effect.

Abstract: The quality of music output from audio systems is improved by simulating the effect of low frequency signals in the human ear. This thus allows listeners to perceive the lower frequency signals, even though the speakers may be incapable of providing such low frequency outputs. A method is provided for processing enhancing bass effect in audio signals. The method also results in the bass enhancement being computationally less intensive. The bass effect enhancement techniques are based on the response of sine and cosine transfer functions and on the directional independence of low frequency components. The human ear is unable to resolve directions from low frequency components. The bass effect enhancement technique alternatively is based on response of an exponential transfer function.