1979results about "Two-channel systems" patented technology

A technique for enhancing audio signals generated from compressed digital audio files is described. The technique uses a Bass Maximizer module, a Harmonic Exciter module and a Quasi Stereo module. The Bass Exciter module enhances the intensity, depth and punch of the bass audio content by creating harmonic sequences from low frequency components contained in the original input signal. The Harmonic Exciter module adds to the treble audio content of the original input signal by generating harmonic series from the high frequency components contained in the input signal. The Quasi Stereo Module creates a stereo image of the enhanced input signal by adding and subtracting delayed and filtered versions of the enhanced input signal with itself to create left and right channeled stereo-like outputs. The technique provides a useful tool to regenerate from an audio signal more pleasant and joyful sounds.

A sound control system and method applicable to an electronic device having a timing unit. A setting module is provided to set predetermined conditions and corresponding sound volume parameters. A time session control module retrieves the sound volume parameter corresponding to a present condition obtained by the timing unit. Then, a corresponding sound output signal to be outputted by a speaker unit connected to the sound control system is set via a sound effects module according to the sound volume parameter and a sound signal around the electronic device that is received and recognized by a sound recognition module. By such arrangement, the sound control system and method allow the electronic device to provide the user with optimal sound effects depending on the environment and the user's preferences.

A parameter transformer generates level parameters, indicating an energy relation between a first and a second audio channel of a multi-channel audio signal associated to a multi-channel loudspeaker configuration. The level parameter are generated based on object parameters for a plurality of audio objects associated to a down-mix channel, which is generated using object audio signals associated to the audio objects. The object parameters have an energy parameter indicating an energy of the object audio signal. To derive the coherence and the level parameters, a parameter generator is used, which combines the energy parameter and object rendering parameters, which depend on a desired rendering configuration.

A media source, including a sending unit that sends out time-stamped media data packets to one or more receiving media sinks. A timestamp of one of the time-stamped media data packets indicates a time of creation of the one of the time-stamped media data packets. The media source also includes a determining unit that determines a play-out time offset. The media source additionally includes a transmission unit that sends out the play-out time offset to the one or more receiving media sinks once for all time-stamped media data packets of a session.

A multi-dimensional audio processor receives as an input either a single channel signal or a two channel signal from an audio signal source; for example a musical instrument or an audio mixer. The processor is programmable to divide the input among at least 3 output channels in a user-defined manner. The processor is also user programmable to provide a variety of effect and mixing functions for the output channel signals.

A musical jukebox is disclosed which provides for: fast archiving of songs; a flexible user interface; easy and convenient entry of, access to and/or display of data relating to songs archived by the jukebox; easy and convenient search and locate capabilities for locating, reviewing, retrieving and/or playing songs stored in the jukebox; and low cost relative to the functionality, features, conveniences and user-friendliness provided by the jukebox. Fast of flush archiving of songs (as well as other data and signals) is accomplished by first saving sets of data without compression, which allows the data to be entered quickly, and then, compressing later at an appropriate time. A unique MP3 bit allocation encoding scheme is used to compress data. A unique memory allocation supports fast data archiving. The user interface employs two-way communication between a remote control and the jukebox. A searchable song database is structured to enable very fast searching by music category, and also by title and artist. The jukebox is provided with an on-board song track database to automatically identify new songs input to the jukebox.

The present invention describes techniques that can be used to provide novel methods of spatial audio rendering using adapted M-S matrix shuffler topologies. Such techniques include headphone and loudspeaker-based binaural signal simulation and rendering, stereo expansion, multichannel upmix and pseudo multichannel surround rendering.

PCT No. PCT/DK95/00089 Sec. 371 Date Dec. 27, 1996 Sec. 102(e) Date Dec. 27, 1996 PCT Filed Feb. 27, 1995 PCT Pub. No. WO95/23493 PCT Pub. Date Aug. 31, 1995A method and apparatus for simulating the transmission of sound from sound sources to the ear canals of a listener encompasses novel head-related transfer functions (HTFs), novel methods of measuring and processing HTFs, and novel methods of changing or maintaining the directions of the sound sources as perceived by the listener. The measurement methods enable the measurement and construction of HTFs for which the time domain descriptions are surprisingly short, and for which the differences between listeners are surprisingly small. The novel HTFs can be exploited in any application concerning the simulation of sound transmission, measurement, simulation, or reproduction. The invention is particularly advantageous in the field of binaural synthesis, specifically, the creation, by means of two sound sources, of the perception in the listener of listening to sound generated by a multichannel sound system. It is also particularly useful in the designing of electronic filters used, for example, in virtual reality systems, and in the designing of an "artificial head" having HTFs that approximate the HTFs of the invention as closely as possible in order to make the best possible representation of humans by the artificial head, thereby making artificial head recordings of optimal quality.

An information presentation apparatus is provided presenting an interface suitable for selecting desired information from a plurality of information. The apparatus has three methods of presenting information. The first method is to present information by sequentially playing and presenting a plurality of audio information related to the information in turn. The second method is to present the plurality of audio information at the same time wherein each sound has its directivity. The third method is a combination of the first and second methods wherein user can select desired method between the two depending on kinds of information presented.

A panner is provided that incorporates a surround sound decoder. The panner takes as input the desired panning effect that a user requests, separates sounds using surround sound decoding, and places the separated sounds in the desired places in an output sound field.

Methods and apparatus are disclosed for processing an audio sound source to create four-dimensional spatialized sound. A virtual sound source may be moved along a path in three-dimensional space over a specified time period to achieve four-dimensional sound localization. The various embodiments described herein provide methods and systems for converting existing mono, 2-channel and/or multi-channel audio signals into spatialized audio signals have two or more audio channels. The incoming audio signals may be down-mixed, up-mixed or otherwise translated into fewer, greater or the same number of audio channels. The various embodiments also describe methods, systems and apparatus for generating low frequency effect and center channel signals from incoming audio signals having one or more channels.

A tinnitus masking system for use by a person having tinnitus The system comprises a sound delivery system having left and right ear-level audio delivery devices and is configured to deliver a masking sound to the person via the audio delivery devices such that the masking sound appears to originate from a virtual sound source location that substantially corresponds to the spatial location in 3D auditory space of the source of the tinnitus as perceived by the person. The masking sound being represented by left and right audio signals that are converted to audible sound by the respective audio delivery devices.

Techniques and systems for enhancing delivered audio signals are disclosed which may be employed in a delivery system at a server side, a client side, or both. The techniques include forming a processed audio signal by processing audio signals through multiple pathways which operate on different frequency bands using dynamic processing and other elements, and thereafter providing recording or listening environment enhancements and other sound enhancements to the processed audio signal. Also disclosed are techniques and systems for implementing the multi-pathway processing and environmental and sound enhancements.

An apparatus for synthesizing a rendered output signal having a first audio channel and a second audio channel includes a decorrelator stage for generating a decorrelator signal based on a downmix signal, and a combiner for performing a weighted combination of the downmix signal and a decorrelated signal based on parametric audio object information, downmix information and target rendering information. The combiner solves the problem of optimally combining matrixing with decorrelation for a high quality stereo scene reproduction of a number of individual audio objects using a multichannel downmix.

A headset having game, chat and microphone audio signals is provided with a programmable signal processor for individually modifying the audio signals and a memory configured to store a plurality of user-selectable signal-processing parameter settings that determine the manner in which the audio signals will be altered by the signal processor. The parameter settings collectively form a preset, and one or more user-operable controls can select and activate a preset from the plurality of presets stored in memory. The parameters stored in the selected preset can be loaded into the signal processor such that the sound characteristics of the audio paths are modified in accordance with the parameter settings in the selected preset.

A sound measuring apparatus for measuring a sound-arrival delay time from a speaker to a microphone on the basis of a result obtained by outputting a test signal from the speaker and picking up the test signal using the microphone includes the following elements. A control unit performs control so that the test signal is expanded in a time axis and is then output from the speaker. A delay time measuring unit measures an expansion-based measured delay time on the basis of a delay time that is measured on the basis of a time difference between the test signal expanded in the time axis and output from the speaker and a signal obtained from the microphone by picking up the output expanded test signal, and obtains the sound-arrival delay time as the expansion-based measured delay time.

An electronic apparatus is provided. The electronic apparatus includes a sensor to sense an orientation of the electronic apparatus, an audio processor to divide a stereo audio signal into a left audio signal and a right audio signal, a plurality of speakers disposed at separate locations on a main body of the electronic apparatus to output the left audio signal and the right audio signal, a switching unit to output the left audio signal and the right audio signal from the audio processor to the plurality of speakers, and a controller to control the switching unit to switch the output of the left audio signal and the right audio signal between the plurality of speakers based on a change in the orientation of the electronic apparatus.

A device and method for providing augmented reality enhanced audio. The device is portable and may include a location processing module capable of receiving location information and determining a location of the device in an environment, an orientation processing module capable of determining an orientation of the device at the location, a processor capable of modifying audio content based on the location of the device in the environment and the orientation of the device at the location, and an audio output device capable of outputting the modified audio content. The orientation of the device includes an azimuth and an elevation of the device.

An audio device and an audio processing method are provided for adjusting the position of a virtual speaker. The audio device comprises a decoder which has audio data provided thereto, the audio data including an audio component for a center speaker and a plurality of audio components corresponding to other speakers disposed with the center speaker interposed therewith, and which decodes these audio components to separate them from the audio data, a center delay processor for delaying the audio component for the center speaker received from the decoder, and a downmixing processor for distributing the delayed center speaker audio component between the other speakers and for merging the audio component distributed to each of the other speakers and the original audio component for each other speaker. Audio sounds corresponding to the downmixed audio components are produced from the other speakers.

This invention is a method for binaural localization using a cascade of resonators and anti-resonators to implement an HRTF (head-related transfer function). The spectrum of the cascade reproduces the magnitude spectrum of a desired HRTF. The proposed method provides a considerably more computationally efficient implementation of HRTF filters with no detectable deterioration of output quality while saving memory when storing a large quantity of HRTFs due to the parameterization of its resonators and anti-resonators. Finally, the method offers additional flexibility since the resonators and anti-resonators can be manipulated individually during the design process, making it possible to interpolate smoothly between HRTFs, reduce spectral coloring or achieve higher accuracy at perceptually relevant frequency regions. These HRTF are useful in stereo enhancement and multi-channel virtual surround simulation.

Binaural rendering a multi-channel audio signal into a binaural output signal is described. The multi-channel audio signal has a stereo downmix signal into which a plurality of audio signals are downmixed, and side information having a downmix information, as well as object level information of the plurality of audio signals and inter-object cross correlation information. Based on a first rendering prescription, a preliminary binaural signal is computed from the first and second channels of the stereo downmix signal. A decorrelated signal is generated as an perceptual equivalent to a mono downmix of the first and second channels of the stereo downmix signal being, however, decorrelated to the mono downmix. Depending on a second rendering prescription, a corrective binaural signal is computed from the decorrelated signal and the preliminary binaural signal is mixed with the corrective binaural signal to obtain the binaural output signal.

An apparatus and method of reproducing a 2-channel virtual sound while dynamically controlling a sweet spot and crosstalk cancellation are disclosed. The method includes: receiving broadband signals, setting compensation filter coefficients according to response characteristics of bands and setting stereophonic transfer functions according to spectrum analysis; down mixing an input multi-channel signal into two channel signals by adding head related transfer functions (HRTFs) measured in a near-field and a far-field to the input multi-channel signal, canceling crosstalk of the down mixed signals on the basis of compensation filter coefficients calculated using the set stereophonic transfer functions, and compensating levels and phases of the crosstalk cancelled signals on the basis of the set compensation filter coefficients for each of the bands.

A desired acoustic signal is extracted from a noisy environment by generating a signal representative of the desired signal with processor (30). Processor (30) receives aural signals from two sensors (22, 24) each at a different location. The two inputs to processor (30) are converted from analog to digital format and then submitted to a discrete Fourier transform process to generate discrete spectral signal representations. The spectral signals are delayed to provide a number of intermediate signals, each corresponding to a different spatial location relative to the two sensors. Locations of the noise source and the desired source, and the spectral content of the desired signal are determined from the intermediate signal corresponding to the noise source locations. Inverse transformation of the selected intermediate signal followed by digital to analog conversion provides an output signal representative of the desired signal with output device (90). Techniques to localize multiple acoustic sources are also disclosed. Further, a technique to enhance noise reduction from multiple sources based on two-sensor reception is described.

A frequency-domain method for format conversion or reproduction of 2-channel or multi-channel audio signals such as recordings is described. The reproduction is based on spatial analysis of directional cues in the input audio signal and conversion of these cues into audio output signal cues for two or more channels in the frequency domain.

A new approach to tracking head motion for headphone-based sound is described. Called MTB2.0 for “Motion-Tracked Binaural with 2 Channels”, the method may be used for any 2-channel binaural signals without any increase in the bandwidth requirement. MTB2.0 provides a simple and effective means to improve the quality of headphone-based sound reproduction by sensing the orientation of the listener's head and using the sensed orientation to appropriately modify the signals sent to the two ears. MTB2.0 method increases the realism and removes some of the shortcomings of binaural sound capture and recording, as well as improves the quality of binaural rendering of stereo.