Abstract: The MPEG2 Advanced Audio Coder (AAC) standard limits the number of filters used to either one filter for a “short” block or three filters for a “long” block. In cases where the need for additional filters is present but the limit of permissible filters has been reached, the remaining frequency spectra are simply not covered by TNS. Two solutions are proposed to deploy TNS filters in order to get the entire spectrum of the signal into TNS. The first method involves a filter bridging technique and complies with the current AAC standard. The second method involves a filter clustering technique. Although the second method is both more efficient and accurate in capturing the temporal structure of the time signal, it is not AAC standard compliant. Thus, a new syntax for packing filter information derived using the second method for transmission to a receiver is also outlined.

Abstract: The invention relates to an audio processing device (1) including: at least one connector (2, 7), suitable for receiving data defining an input sound signal; a processing module (5, 6) suitable for determining at least one signal intended for a loudspeaker (12, 13) as a function of said defining data, a communication module (8, 9) for communicating with at least one other audio processing device, according to claim 1; said device being characterized in that it furthermore includes a detection module (4) suitable for detecting the proximity of another device according to claim 1 and in that the processing module is suitable for determining said signal intended for the loudspeaker as a function furthermore of a number of other detected device(s).

Abstract: A context-aware voice guidance method is provided that interacts with other voice services of a user device. The voice guidance does not provide audible guidance while the user is making a verbal request to any of the voice-activated services. Instead, the voice guidance transcribes its output on the screen while the verbal requests from the user are received. In some embodiments, the voice guidance only provides a short warning sound to get the user's attention while the user is speaking on a phone call or another voice-activated service is providing audible response to the user's inquires. The voice guidance in some embodiments distinguishes between music that can be ducked and spoken words, for example from an audiobook, that the user wants to pause instead of being skipped. The voice guidance ducks music but pauses spoken words of an audio book in order to provide voice guidance to the user.

Abstract: In a vehicle equipped with a vehicle sound system, determining a virtual fixed gear ratio; determining a virtual RPM based on the virtual fixed gear ratio; generating a set of harmonic signals based on the virtual RPM, the harmonic signals being sine-wave signals proportional to harmonics of the virtual RPM; processing the harmonic signals to produce a set of processed harmonic signals; and in the vehicle sound system, transducing the processed harmonic signals to acoustic energy thereby to produce an engine sound within a passenger cabin of the vehicle.

Abstract: An earpiece device (10, 110, 210, 210?) provides advanced communication to the user thereof, for controlling a total sound dose (TND), including an audio output from an audio source (16) reaching an ear of the user occluded at the outer ear canal (15) by an in-ear device (14), for allowing an external ambient sound to be heard by the ear of the user, and for allowing a voice (NSV) of the user to be transmitted to a telecommunication transmission link (TTL). The present invention also contemplates a method of operation of the earpiece device (10, 110, 210, 210?).

Abstract: A hearing aid compatible portable electronic audio device is configured to automatically determine whether or not the device is being used by a hearing impaired user who is wearing a hearing aid, and select a mode of operation based on this determination. The device includes a proximity sensor and a magnetic field sensor. The proximity sensor is used to detect a change in distance of the device to the user's ear. The magnetic field sensor is used to detect a change in magnetic field caused by the device moving relative to the hearing aid. The device selects between a normal audio mode of operation and a hearing aid compatible mode of operation based on both the change in detected distance and the change in detected magnetic field. Other embodiments are also described and claimed.

Abstract: A multi-channel and multi-zone audio environment is provided. Various inventions are disclosed that allow playback devices on one or more networks to provide an effective multi-channel and a multi-zone audio environment using timing information. According to one example, timing information is used to coordinate playback devices connected over a low-latency network to provide audio along with a video display. In another example, timing information is used to coordinate playback devices connected over a mesh network to provide audio in one or more zones or zone groups.

Abstract: A telephone includes at least two microphones and a circuit for processing audio signals coupled to the microphones. The circuit processes the signals, in part, by providing at least one statistic representing maximum normalized cross-correlation of the signals from the microphones, doaEst, dirGain, or diffGain and comparing the at least one statistic with a threshold for that statistic. At least one of noise reduction and speech enhancement is controlled by an indication of near-field sounds in accordance with the comparison. Indication of near-field speech can be further enhanced by combining statistics, including a statistic representing inter-microphone level difference, each of which have their own threshold. dirGain and diffGain are derived from signals incident upon the microphones such that the desired near-field signal is not suppressed.

Abstract: A reproducing apparatus is disclosed. The reproducing apparatus includes a reproducing section, a notifying section, and a notification controlling section. The reproducing section reproduces content data. The notifying section notifies a user that there are content data correlated with content data reproduced by the reproducing section. The notification controlling section determines whether content data correlated with content data reproduced by the reproducing section have been retrieved and controls the notifying section to notify the user that there are the correlated content data when the correlated content data have been retrieved.

Abstract: A signal processing device includes a noise analysis unit for analyzing a frequency component of a noise signal obtained by converting a collected sound into an electrical signal, a plurality of filtering units for carrying out predetermined filtering operations on the noise signal on the basis of an analysis result, and an output control unit for temporally varying a synthesis rate of outputs of the plurality of filtering units according to a change in the analysis result of the noise analysis unit. When the analysis result of the noise analysis unit changes, one filtering unit starts a predetermined filtering operation by characteristics different from those of other filtering units that carry out predetermined filtering operations on the noise signal according to the change in the analysis result of the noise analysis unit.

Abstract: An acoustic system includes: a first customer-side microphone; a first counselor-side microphone; a first sound changing unit; a first loudspeaker; a second customer-side microphone; a second counselor-side microphone; and a second loudspeaker. Between the second loudspeaker and the first customer-side microphone, a first sound transmission path is provided. Also, between the second loudspeaker and the first counselor-side microphone, a second sound transmission path is provided. These sound transmission paths have substantially the same length. The first sound signal generated by the first customer-side microphone is made to have a phase substantially opposite to that of the second sound signal generated by the first counselor-side microphone, and the sound signals are added together.

Abstract: A pre-distortion system for improved mobile device communications via cancellation of nonlinear distortion is disclosed. The pre-distortion system may transmit an acoustic signal from a network to a device, wherein the acoustic signal includes a linear signal and a nonlinear cancellation signal that cancels at least a portion of nonlinear distortions created once a loudspeaker in the device emits the linear signal. Thus, when a loudspeaker of a mobile device is operating and nonlinear distortions are generated by the loudspeaker or adjacent components of the mobile device in close proximity to the loudspeaker, the pre-distortion system may create one or more nonlinear cancellation signals in the network. The nonlinear cancellation signal may be combined with the linear signal sent to the loudspeaker to cancel the nonlinear distortion signal created by the loudspeaker emitting acoustic sounds from the linear signal. Thus, the nonlinear cancellation signal becomes a pre-distortion signal.

Abstract: In order to obtain coded data which does not strike viewers and listeners as being incongruous, when plural audio data are to be coded, a coding program groups the respective audio data into one audio data, codes the grouped audio data in sequence with a predetermined number of samples being treated as units, and sets delimitations corresponding to the delimitations of the plural audio data in the coded data at coding units of the coded data.

Abstract: A mobile device-based stethoscope system that transmits, records, and analyzes sounds to generate a list of matching conditions and facilitates easy attachment across various electronic medical record platforms and other means of communication. The invention is configured to allow the use of either an integrated wireless stethoscope, or an in-line adapter for a conventional stethoscope. Patient sounds are sent from the selected stethoscope head to the mobile device having a software application that allows for the analysis, attachment, and further manipulation of the data.

Abstract: An information distribution system enhances a vehicle audio/video system. The system flexibly delivers audio and video, potentially obtained from many different sources, to different locations in the vehicle. Each passenger receives the audio or video that is of interest to them without distracting interruptions.

Abstract: A signal analysis device includes: a signal reception unit which receives an input signal containing a plurality of constituent elements; and a signal analysis unit which generates analysis information indicating the relationship between the constituent elements from the input signal.

Abstract: The application relates to audio encoder and decoder systems. An embodiment of the encoder system comprises a downmix stage for generating a downmix signal and a residual signal based on a stereo signal. In addition, the encoder system comprises a parameter determining stage for determining parametric stereo parameters such as an inter-channel intensity difference and an inter-channel cross-correlation. Preferably, the parametric stereo parameters are time- and frequency-variant. Moreover, the encoder system comprises a transform stage. The transform stage generates a pseudo left/right stereo signal by performing a transform based on the downmix signal and the residual signal. The pseudo stereo signal is processed by a perceptual stereo encoder. For stereo encoding, left/right encoding or mid/side encoding is selectable. Preferably, the selection between left/right stereo encoding and mid/side stereo encoding is time- and frequency-variant.

Abstract: A tone generation system includes a square wave signal generator configured to generate a first series of square wave signals to replicate a fundamental frequency of a desired mechanical tone, and a second series of square waves signals to replicate a second harmonic of the desired mechanical tone. An amplifier is configured to receive the first and second series of square waves, and a speaker is connected to receive an output signal from the amplifier.

Abstract: The invention relates to a method for improving a stereo audio signal of an FM stereo radio receiver. The method comprises determining one or more parametric stereo parameters based on the stereo audio signal in a frequency-variant or frequency-invariant manner. Preferably, these PS parameters are time- and frequency-variant. Moreover, the method comprises generating the improved stereo signal based on a first audio signal and the one or more parametric stereo parameters. The first audio signal is obtained from the stereo audio signal, e.g. by a downmix operation.

Abstract: A headphone driver includes a pre-main amplifier and a main amplifier. The pre-main amplifier receives an input signal from first and second node and outputs the input signal to a third node. The main amplifier is between the third node and a fourth node. A first switch is between the fourth node and a fifth node. A second switch is between the fourth node and a sixth node. A third switch is between the fourth node and a seventh node. A fourth switch is between the fifth and seventh nodes. A fifth switch is between the first and fifth nodes. A capacitor is between the third and fifth nodes. A first feedback resistor is between the first and seventh nodes. A second feedback resistor is between the second and sixth nodes, and the sixth and seventh nodes are connected to a speaker.