Abstract: An audio device integration system is provided. One or more after-market audio devices, such as a CD player, CD changer, MP3 player, satellite receiver, DAB receiver, or the like, is integrated for use with an existing OEM or after-market car stereo system, wherein control commands can be issued at the car stereo and responsive data from the audio device can be displayed on the stereo. Control commands generated at the car stereo are received, processed, converted into a format recognizable by the audio device, and dispatched to the audio device for execution. Information from the audio device, including track, disc, song, station, time, and other information, is received, processed, converted into a format recognizable by the car stereo, and dispatched to the car stereo for display thereon. One or more auxiliary input sources can be integrated with the car stereo, and selected between using the controls of the car stereo.

Abstract: In a method for canceling unwanted signals from at least one external sound source, such as a loudspeaker, by means of headphones provided with microphones, at least known sound signals from the at least one external sound source are compensated by anti-phase sound signals. These sound signals simulate the at least known sound signals from said at least one external sound source in anti-phase. Said anti-phase sound signals are generated in the headphones in response to signals derived from audio input signals of the at least one external sound source in a filter device which is controlled by the resulting microphone signals.

Abstract: A vehicle seatback audio controller includes an entertainment interface that receives an entertainment audio signal from entertainment audio source, and a telematics interface that receives a telematics audio signal from a telematics audio source. The controller has an output that provides either the telematics audio signal to a seatback speaker when the telematics audio source is active or the entertainment audio signal to the seatback speaker when the telematics audio source is inactive. The controller may include an accelerometer or a microphone that generates a vibration signal and/or a noise signal. A level detector may receive the vibration signal and/or a noise signal, and adjust a level of the telematics audio signal to the seatback speaker as a function of the vibration signal and/or a noise signal.

Abstract: The present invention is related to a method for adjusting the frequency characteristic of a digital audio processor having adjustable parameters including at least low frequency gain, high frequency gain and volume gain. The invention is also related to an audio device executing the inventive method every time the characteristic frequency response curve is adjusted. Today's audio devices frequently offer the feature that the user can select among a group of different sound characteristics. Typically the user may want to adapt the sound characteristic of the device to the kind of music he is listening, e.g. rock, jazz, or classic music. In technical terms the sound characteristic is widely determined by the characteristic frequency response curve of the audio signal processing. Those characteristic curves are essentially defined by a gain value in the low, middle and high frequency range. Normally, a 3-band audio processor, which allows adjusting these three parameters independently is used for this purpose.

Abstract: A single folded, expanding horn loudspeaker reproduces low frequency audible sound at high power output levels. A compact enclosure houses a plurality of identical transducers, characterized by small vibrational surface areas. The throats for each transducer into the horn are acoustically differentially spaced from the mouth of the horn. Transducer drive circuitry adjusts a drive signal for the transducers to compensate for the different impedance and propagation time to the horn mouth at each throat.

Abstract: A vehicle-mounted three dimensional sound signal reproducing•silencing unit includes at least two speakers disposed for each of seats to be occupied by listeners so as to generate a three dimensional sound field space in a region inclusive of left and right external ears of each of the listeners, and three dimensional signal processing means for converting an output signal from a sound source into a three dimensional sound field signal for supplying it to each of the speakers, thereby reproducing a three dimensional sound field with rich presence for each of the seats. The unit also includes antiphase signal generating means for supplying a silencing signal to the three dimensional signal processing means, thereby silencing sounds for each of the seats without giving an effect on the other seats.

Abstract: An adaptive noise reduction method and apparatus capable of reducing efficiently variable period noise from a main input is provided. The pitch of a noise waveform to be reduced can be variable with a change in a period of motor noise occurring when the revolution period is changed by disc motor control of DVD-RAM, by revolution speed control of other motors, and revolution period change on starting a motor and the like. Therefore, the renewal of adaptive filter coefficient becomes almost unnecessary, thus allowing noise reduction to be performed without degrading a noise canceling effect.

Abstract: Methods for generating a directional sound environment include providing a headgear unit having a plurality of microphones thereon. A sound signal is detected from the plurality of microphones. A transfer function is applied to the sound signal to provide a transformed sound signal, and the transformed sound signal provides an approximation of free field hearing sound at a subject's ear inside the headgear unit.

Abstract: A system and method for enhancing song-to-song transitions in a multi-channel audio environment, preferably a surround environment. In the inventive method, by independently manipulating the volumes of the various channels of each program during transitions, an illusion of motion is imparted to the program which is ending to create an impression that the song is exiting while motion is imparted to the program which is starting to create an impression that the song is entering. Optionally delay may be used to further enhance the effect. In a preferred system, each audio channel includes inputs for both the ending and starting programs, independent volume controls for each program. In one preferred embodiment, the system further includes a means for imparting delay to each program and volume control for the delayed audio to provide greater depth to the illusion of motion.

Abstract: An analog-digital interface device is provided, which is configured to be housed within or removably coupled with an analog mixer, to thereby form a digital-output analog mixer. The analog-digital interface device includes analog-digital conversion circuitry and digital transmission circuitry that convert analog audio from the analog mixer to digital format for transmission along a digital transport interface to an external digital computing device, such as a PC or MAC computer. In one embodiment, the analog audio can be tapped from any one or more points along a signal path of the analog mixer and sent to the analog-digital interface device. In another embodiment, the analog-digital interface device is further configured to receive digital signals back from the external digital computing device, convert them to analog signals, and return them to the analog mixer at any one or more points along a signal path of the analog mixer.

Abstract: The present invention relates to a wave-filtering device with an adjustable frequency bandwidth installed in an output unit of an integrated system. The wave-filtering device is used for receiving a first output signal comprising an alternating current signal and a direct current bias voltage, and comprises a direct current bias voltage detector for receiving the first output signal and obtaining a control voltage according to the direct current bias voltage. A low-pass wave filter is electrically connected to the direct current bias voltage detector and receives the control voltage so as to adjust the frequency bandwidth. The low-pass wave filter is used for receiving the first output signal and obtaining a second output signal to be the output of the wave-filtering device with adjustable frequency bandwidth. The low-pass wave filter comprises a variable resistance the value of which varies in accordance with the value of the control voltage.

Abstract: The present invention relates to a method for encoding an audio signal. In a first embodiment a model relating to temporal masking of sound provided to a human ear is provided. A temporal masking index is determined in dependence upon a received audio signal and the model using a forward and a backward masking function. Using a psychoacoustic model a masking threshold is determined in dependence upon the temporal masking index. Finally, the audio signal is encoded in dependence upon the masking threshold. The method has been implemented using the MPEG-1 psychoacoustic model 2. Semiformal listening test showed that using the method for encoding an audio signal according to the present invention the subjective high quality of the decoded compressed sounds has been maintained while the bit rate was reduced by approximately 10%. In a second embodiment, the inharmonic structure of audio signals is modeled and incorporated into the MPEG-1 psychoacoustic model 2.

Abstract: An audio-signal-processing apparatus includes a band-decomposition unit utilizing a band-decomposition characteristic based on the harmonic-tone composition of musical instruments. A low interval limit generates the harmonic-tone to each band signal after decomposing the bass signal into a plurality of band signals. The audio-signal processing apparatus generates a harmonic-tone individually for each pure tone component, which includes complex tones and suppresses deterioration of the sound quality.

Abstract: There is provided a signal processing apparatus that is capable of giving terminal names which are the most suitable for processing on signals. A plurality of kinds of signals are externally input via a plurality of input terminals of an input terminal group. Processing is performed on the plurality of kinds of signals to generate a signal for at least one channel. The signal for the at least one channel is externally output via a plurality of output terminals of an output terminal group. A plurality of effective terminals included in at least one of the input terminal group and the output terminal group are automatically numbered.

Abstract: The present invention implements a method and an apparatus for retrieving a sound data desired by the user on the basis of its subjective impression over the sound data. The subjective impression on the desired sound data is entered by the user and converted to a numerical value. A target sound impression value which is a numerical form of the impression on the sound data is calculated from the numerical value. The target sound impression value is then used as a retrieving key for accessing a sound database where the audio signal and the sound features of a plurality of the sound data are stored. This allows the desired sound data to be retrieved on the basis of the subjective impression of the user on the sound data.

Abstract: Enhanced audio communications is provided in an interactive environment, such as a gaming environment, having multiple participants. An input audio signal, such as a speech signal, that is originated by one of the multiple participants is received. In addition, one or more instructions that correspond to the input audio signal are received. Based on the one or more instructions, an output audio signal is generated from the input audio signal. The output audio signal is designated for transmission to one or more of the participants.

Abstract: Systems, devices and methods are provided for diotically presenting second-order gradient directional hearing aid signals. The present subject matter provides an improved signal-to-noise ratio, and presents a desired directional signal to each ear. One aspect is a hearing aid system. In one embodiment, the system includes a first microphone system in a first device and a second microphone system in a second device. The first microphone system has a first output signal, and the second microphone system has a second output signal. Each output signal includes a first-order directional signal. The system further includes a first receiver circuit and a second receiver circuit. The combination of the first output signal and the second output signal provides a diotic presentation of a second-order gradient signal to both the first receiver circuit and the second receiver circuit. Other aspects are provided herein.

Abstract: In the present invention, in parallel with a resistor which determines an amplification rate being fed back from an output to an input of an output stage OP amp in an audio amplifier circuit, providing a first diodes connected bi-directionally together with a first switch circuit, the output voltage of the output stage OP amp is limited to a voltage, when the first diodes are turned ON. The limit voltage of the first diodes corresponds to a speaker having a low withstand voltage. Accordingly, through ON/OFF of the first switch circuit, a power output from the output stage OP amp can be selected depending on the withstand voltage of a speaker to be connected.

Abstract: An electronic control that monitors the switching means used to operate transducers to detect conditions that can cause improper transducer operation. In the application the unenergized state of the transducer is always safe. Should a switch in the transducer circuitry be in an erroneous state, the control uses another switch to prevent improper operation of the transducer without requiring the involvement of the operator. Monitoring enables the control to prevent transducer operation before failures allow hazardous operation. Any switching means, activated by the control, an override, or both may be monitored as long as its intended state is known to the control. The control can also open the transducer circuitry when an externally controlled switch in the transducer circuitry changes state at a destructive rate.

Abstract: The invention relates to a method for canceling feedback in an acoustic system comprising a microphone, a signal path, a speaker and means for detecting presence of feedback between the speaker and the microphone, the method comprising providing a LMS algorithm for processing the signal; where the LMS algorithm operates with a predetermined adaptation speed when feedback is not present; where the LMS algorithm operates an adaptation speed faster than the predetermined adaptation speed when feedback is present, and where the means for detecting the presence of feedback is used to control the adaptation speed selection of the LMS algorithm. The invention further relates to a method comprising using a highpass filter to prevent low-frequency signals from entering the LMS algorithm. The invention still further relates to a method comprising using bandwidth detection means for determining the presence of a feedback signal. The invention also relates to hearing aids for implementing these methods.