Patent application number | Description | Published |
20080199014 | Low power downmix energy equalization in parametric stereo encoders - 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. | 08-21-2008 |
20100169102 | LOW COMPLEXITY MPEG ENCODING FOR SURROUND SOUND RECORDINGS - 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. | 07-01-2010 |
20100208899 | METHOD AND SYSTEM FOR ENHANCING BASS EFFECT IN AUDIO SIGNALS - 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. | 08-19-2010 |
20110099021 | CONTENT FEATURE-PRESERVING AND COMPLEXITY-SCALABLE SYSTEM AND METHOD TO MODIFY TIME SCALING OF DIGITAL AUDIO SIGNALS - A time-domain system and method of modifying the time scale of digital audio signals includes a pre-processor. The pre-processor forms a synthesized signal for processing with minimum computation and that has optional features to give preference to certain audio channels and/or frequency bands, a mechanism of adaptively characterizing the temporal features of the synthesized signal by its normalized power and zero-crossing count, and a mechanism of identifying a segment of the synthesized signal where the time scale can be modified without introducing artifacts or losing content. | 04-28-2011 |
20110150242 | ADAPTIVE LOUDNESS LEVELLING FOR DIGITAL AUDIO SIGNALS - 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. The target gain to maintain the audio signal to the desired loudness level is calculated by a volume leveller, regulated by a gain controller that simulates the temporal masking effect to get rid of unnecessary gain fluctuations, ensuring a pleasant sound. | 06-23-2011 |
20110176696 | METHOD AND SYSTEM FOR ENHANCING BASS EFFECT IN AUDIO SIGNALS - 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. | 07-21-2011 |
20110216906 | ENABLING 3D SOUND REPRODUCTION USING A 2D SPEAKER ARRANGEMENT - 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. | 09-08-2011 |
20120035936 | INFORMATION REUSE IN LOW POWER SCALABLE HYBRID AUDIO ENCODERS - 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). | 02-09-2012 |
20120163622 | NOISE DETECTION AND REDUCTION IN AUDIO DEVICES - 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. | 06-28-2012 |
20130044894 | SYSTEM AND METHOD FOR EFFICIENT SOUND PRODUCTION USING DIRECTIONAL ENHANCEMENT - 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. | 02-21-2013 |
20130148814 | AUDIO ACQUISITION SYSTEMS AND METHODS - Audio acquisition systems and methods to determine a direction of arrival of an audio signal are disclosed. In an embodiment, an apparatus includes a continuous sampling stage configured to receive audio information and to generate one or more correlations from the received audio information, and a processing stage configured to receive the one or more correlations and to generate direction of arrival information for the audio information. In another embodiment, a method includes generating audio signals from an ambient acoustic environment, and performing beamforming on the generated audio signals. The method further includes calculating signal-to-interference ratios from the beamformed signals, forming correlations between the signal-to-interference ratios and audio sampling angles, selecting at least one correlation based upon predetermined selection criteria, and determining a direction of arrival for the audio signals. | 06-13-2013 |
20130170666 | ADAPTIVE SELF-CALIBRATION OF SMALL MICROPHONE ARRAY BY SOUNDFIELD APPROXIMATION AND FREQUENCY DOMAIN MAGNITUDE EQUALIZATION - Methods and apparatus for self-calibration of small-microphone arrays are described. In one embodiment, self-calibration is based upon a mathematical approximation for which a detected response by one microphone should approximately equal a combined response from plural microphones in the array. In a second embodiment, self-calibration is based upon matching gains in each of a plurality of Bark frequency bands, and applying the matched gains to frequency domain microphone signals such that the magnitude response of all the microphones in the array approximates an average magnitude response for the array. The methods and apparatus may be implemented in hearing aids or small audio devices and used to mitigate adverse aging and mechanical effects on acoustic performance of small-microphone arrays in these systems. | 07-04-2013 |
20140128004 | CONVERTING SAMPLES OF A SIGNAL AT A SAMPLE RATE INTO SAMPLES OF ANOTHER SIGNAL AT ANOTHER SAMPLE RATE - In an embodiment, an apparatus includes a determiner, converter, adapter, and modifier. The determiner is configured to generate a representation of a difference between a first frequency at which a first signal is sampled and a second frequency at which a second signal is sampled, and the converter is configured to generate a second sample of the first signal at a second time in response to the representation and a first sample of the first signal at a first time. The adapter is configured to generate a sample of a modifier signal in response to the second sample of the first signal, and the modifier is configured to generate a modified sample of the second signal in response to a sample of the second signal and the sample of the modifier signal. For example, such an apparatus may be able to reduce the magnitude of an echo signal in a system having an audio pickup (e.g., a microphone) near an audio output (e.g., a speaker). | 05-08-2014 |
20140195232 | Methods, systems, and circuits for text independent speaker recognition with automatic learning features - Embodiments provide a method and system of text independent speaker recognition with a complexity comparable to a text dependent version. The scheme exploits the fact that speech is a quasi-stationary signal and simplifies the recognition process based on this theory. The modeling allows the speaker profile to be updated progressively with the new speech sample that is acquired during usage time. | 07-10-2014 |
20140200890 | METHODS, SYSTEMS, AND CIRCUITS FOR SPEAKER DEPENDENT VOICE RECOGNITION WITH A SINGLE LEXICON - Embodiments reduce the complexity of speaker dependent speech recognition systems and methods by representing the code word (i.e., the word to be recognized) using a single Gaussian Mixture Model (GMM) which is adapted from a Universal Background Model (UBM). Only the parameters of the GMM need to be stored. Further reduction in computation is achieved by only checking the GMM component that is relevant to the keyword template. In this scheme, keyword template is represented by a sequence of the index of best performing component of the GMM of the keyword model. Only one template is saved by combining the registration template using Longest Common Sequence algorithm. The quality of the word model is continuously updated by performing expectation maximization iteration using the test word which is accepted as keyword model. | 07-17-2014 |