251 results about "Psychoacoustics" patented technology

A time-discrete audio signal is processed to provide a quantization block with quantized spectral values. Furthermore, an integer spectral representation is generated from the time-discrete audio signal using an integer transform algorithm. The quantization block having been generated using a psychoacoustic model is inversely quantized and rounded to then form a difference between the integer spectral values and the inversely quantized rounded spectral values. The quantization block alone provides a lossy psychoacoustically coded/decoded audio signal after the decoding, whereas the quantization block, together with the combination block, provides a lossless or almost lossless coded and again decoded audio signal in the decoding. By generating the differential signal in the frequency domain, a simpler coder/decoder structure results.

A method for compressing audio input signals to form a master bit stream that can be scaled to form a scaled bit stream having an arbitrarily prescribed data rate. A hierarchical filterbank decomposes the input signal into a multi-resolution time/frequency representation from which the encoder can efficiently extract both tonal and residual components. The components are ranked and then quantized with reference to the same masking function or different psychoacoustic criteria. The selected tonal components are suitably encoded using differential coding extended to multichannel audio. The time-sample and scale factor components that make up the residual components are encoded using joint channel coding (JCC) extended to multichannel audio. A decoder uses an inverse hierarchical filterbank to reconstruct the audio signals from the tonal and residual components in the scaled bit stream.

A cabin communication system for improving clarity of a voice spoken within an interior cabin having ambient noise includes a microphone for receiving the spoken voice and the ambient noise and for converting the spoken voice and the ambient noise into an audio signal, the audio signal having a first component corresponding to the spoken voice and a second component corresponding to the ambient noise, a speech enhancement filter for removing the second component from the audio signal to provide a filtered audio signal, the speech enhancement filter removing the second component by processing the audio signal by a method taking into account elements of psycho-acoustics of a human ear, and a loudspeaker for outputting a clarified voice in response to the filtered audio signal.

An efficient audio coding device that quantizes and encodes digital audio signals with a reduced amount of computation. A spatial transform unit subjects samples of a given audio signal to a spatial transform, thus obtaining transform coefficients of the signal. With a representative value selected out of the transform coefficients of each subband, a quantization step size calculator estimates quantization noise and calculates, in an approximative way, a quantization step size of each subband from the estimated quantization noise, as well as from a masking power threshold determined from a psycho-acoustic model of the human auditory system. A quantizer then quantizes the transform coefficients, based on the calculated quantization step sizes, thereby producing quantized values of those coefficients. The quantization step sizes are also used by a scalefactor calculator to calculate common and individual scalefactors. A coder encodes at least one of the quantized values, common scalefactor, and individual scalefactors.

A system and method of improving signal intelligibility over an interference signal is provided. The system includes a psychoacoustic professor having a psychoacoustic model wherein the level of a signal-of-interest is improved so as to be audible above noise and so as not to exceed a predetermined maximum output level. The system can be combined with active noise cancellation.

Disclosed is an amplitude-scaling resilient audio watermarking apparatus and method. An encoding apparatus includes: a polyphase filterbank for dividing an audio signal into a plurality of subbands; a psychoacoustic module for applying a psychoacoustic model to the audio signal to provide a signal-to-mask ratio; a watermark encoder for evaluating an encoding parameter from the subbands according to the signal-to-mask ratio and embedding the encoding parameter and a watermark into subbands corresponding to the middle frequency; and a synthesis filterbank for synthesizing the subband signals to output a watermarked audio signal. A decoding apparatus includes: a polyphase filterbank for dividing a received audio signal into the predetermined number of subbands; an EM estimator for estimating an scale factor from an encoding parameter contained in the audio signal and a watermarked subband according to an EM algorithm, and generating the quantizer step size Δ_d of a decoder according to the amplitude-scaling; a watermark decoder for extracting a watermark from a subband corresponding to the middle frequency considering the quantizer step size; and an integrated determiner for integrating outputs of the watermark decoder to determine a watermark.

A method of encoding a digital signal, particularly an audio signal, which predicts favorable scalefactors for different frequency subbands of the signal. Distortion thresholds which are associated with each of the frequency subbands of the signal are used, along with transform coefficients, to calculate total scaling values, one for each of the frequency subbands, such that the product of a transform coefficient for a given subband with its respective total scaling value is less than a corresponding one of the distortion thresholds. In an audio encoding application, the distortion thresholds are based on psychoacoustic masking. The invention may use a novel approximation for calculating the total scaling values, which obtains a first term based on a corresponding distortion threshold, and obtains a second term based on a sum of the transform coefficients. Both of these terms may be obtained using lookup tables. The total scaling values can be normalized to yield scalefactors by identifying one of the total scaling values as a minimum nonzero value, and using that minimum nonzero value to carry out normalization. Encoding of the signal further includes the steps of setting a global gain factor to this minimum nonzero value, and quantizing the transform coefficients using the global gain factor and the scalefactors.

An indoor ultrasonic location tracking system that can utilize standard audio speakers to provide indoor ranging information to modern mobile devices like smartphones and tablets. The method uses a communication scheme based on linearly increasing frequency modulated chirps in the audio bandwidth just above the human hearing frequency range where mobile devices are still sensitive. The method uses gradual frequency and amplitude changes that minimize human perceivable (psychoacoustic) artifacts derived from the non-ideal impulse response of audio speakers. Chirps also benefit from Pulse Compression, which improves ranging resolution and resilience to both Doppler shifts and multi-path propagation that plague indoor environments. The method supports the decoding of multiple unique identifier packets simultaneously. A Time-Difference-of-Arrival pseudo-ranging technique allows for localization without explicit synchronization with the broadcasting infrastructure. An alternate received signal strength indicator based localization technique allows less accurate localization at the benefit of sparser transmission infrastructure.

A front surround reproduction system improving the stereo effect of mid and low frequency signals by using a psychoacoustic model, and a method thereof. An audio reproducing system to reproduce multi-channel audio signals by using a plurality of speakers includes a split unit to copy the input multi-channel signals and to split the signals into two groups of multi-channel signals, a virtual sound processing unit to generate a virtual sound signal based on a head related transfer function (HRTF) from the one group of the multi-channel signals split in the split unit, a beam forming processing unit to generate a sound beam signal by adjusting the delays and levels of the multi-channel signals belonging to the other group split in the split unit, and a crossover network unit to adjust the characteristics of the virtual sound signal and the sound beam signal generated in the virtual sound processing unit and the beam forming processing unit, respectively, and to provide the virtual sound signal and the sound beam signal to mid and low frequency speaker arrays and high frequency speaker arrays, respectively.

A method for encoding an audio signal, including providing a masking function, representative of psychoacoustic masking; setting a quality value for data of the encoded signal, adjusting the masking function dependent upon the quality value; and allocating bits for quantization of the encoded signal based on the incremental masking function.

The present invention counterbalances background noise by applying dynamic equalization. A psychoacoustic model representing the perception of masking effects of background noise relative to a desired foreground soundtrack is used to accurately counterbalance background noise. A microphone samples what the listener is hearing and separates the desired soundtrack from the interfering noise. The signal and noise components are analyzed from a psychoacoustic perspective and the soundtrack is equalized such that the frequencies that were originally masked are unmasked. Subsequently, the listener may hear the soundtrack over the noise. Using this process the EQ can continuously adapt to the background noise level without any interaction from the listener and only when required. When the background noise subsides, the EQ adapts back to its original level and the user does not experience unnecessarily high loudness levels.

Methods and apparatus for maximizing speech intelligibility use psycho-acoustic variables of a model of speech perception to control the determination of optimal frequency-band specific gain adjustments. Speech signals (or other audio input) whose intelligibility is to be improved are characterized by parameters which are applied to the model. These include measurements or estimates of speech intensity level, average noise spectrum of the incoming audio signal, and/or the current frequency-gain characteristic of the hearing compensation device. Characterizations of listeners based on hearing test results, for example, may also be applied to the model. Frequency-band specific gain adjustments generated by use of the model can be used for hearing aids, assistive listening devices, telephones, cellular telephones, or other speech delivery systems, personal music delivery systems, public-address systems, sound systems, speech generating systems, or other devices or mediums which project, transfer or assist in the detection or recognition of speech.

An audio processing system may modify an input surround sound signal to generate a spatially equilibrated output surround sound signal that is perceived by a user as spatially constant for different sound pressures of the surround sound signal. The audio processing system may determine based on a psychoacoustic model of human hearing, a loudness and a localisation for a combined sound signal. The loudness and the localisation may be determined by the system for a virtual user located between the front and the rear loudspeakers that has a predetermined head position in which one ear of the virtual user is directed towards one of front or rear loudspeakers and the other ear of the virtual user being directed towards the other of the front or rear loudspeakers. The audio processing system may adapt the front and/or rear audio signal channels based on the determined loudness and localisation.

Speech enhancement based on a psycho-acoustic model is disclosed that is capable of preserving the fidelity of speech while sufficiently suppressing noise including the processing artifact known as “musical noise”.

A system is disclosed where the well-known psycho-acoustic masking effect, i.e. frequency and/or temporal masking is applied to reduce the human perception of a noise signal. An input signal, such as music or another entertainment signal, is adjusted based on the intensity of the auditory noise by applying existing knowledge about the properties of the human auditory perception and is provided to the human user as a masking sound signal, so that the masking sound elevates the human auditory perception threshold for at least some of the noise signal, whereby the user's perception of that part of the noise signal is reduced or eliminated. The masking sound may be combined with active noise control where a sound in anti-phase with the noise signal is provided for further reduction of the human perception of the noise signal.

The Cellular Automata Music Generator (CAMG) is a software platform, application and engine that generates and modifies musical compositions in real-time using a system of networked modules that utilize the 1-dimensional cellular automata (CA) mathematical model. The music compositions are non-looping (i.e. the same few bars of music do not repeat over and over), can be any length, can have low memory and processing requirements compared to looping WAV or MP3 files, and follow psychoacoustic principles. The music generated is ever evolving and can be globally altered based on one-parameter change. The music engine can be utilized in computer games or to generate unique ringtones or any musical composition for cell phones or other mobile devices based on ten-digit inputs, such as telephone numbers. The ringtones or musical pieces can be generated automatically based on the telephone number of the in-coming telephone call, or manually by the user, who inputs a selected numerical sequence. It can also be used as an audio caller identification system.

The present invention is generally directed to a system and method for enhancing speech using a multi-channel noise filtering process that is based on psychoacoustic masking effects. A speech enhancement/noise reduction scheme according to the present invention is designed to satisfy the psychoacoustic masking principle and to minimize the signal total distortion by exploiting multiple microphone signals to enhance the useful speech signal at reduced level of artifacts.