177 results about "Audio Signal Processor" patented technology

An audio signal decorrelator for deriving an output audio signal from an input audio signal has a frequency analyzer for extracting from the input audio signal a first partial signal descriptive of an audio content in a first audio frequency range and a second partial signal descriptive of an audio content in a second audio frequency range having higher frequencies compared to the second audio frequency range. A partial signal modifier for modifies the first and second partial signals, to obtain first and second processed partial signals, so that a modulation amplitude of a time variant phase shift or time variant delay applied to the first partial signal is higher than that applied to the second partial signal, or for modifying only the first partial signal. A signal combiner combines the first and second processed partial signals, or combines the first processed partial signal and the second partial signal, to obtain an output audio signal.

Described is a unit for amplifying and processing audio signals for driving an electro-acoustic transducer (D), comprising: an input for audio signals; a processor for audio signals (107); an output for a signal for driving said electro-acoustic transducer; and an input for at least one operating quantity of the electro-acoustic transducer. The audio-signal processor is programmed for setting a series of parameters defining a transducer to be emulated, the parameters of which define a model of the transducer to be emulated. The input audio signal is processed on the basis of said at least one operating quantity of the electro-acoustic transducer to obtain a behavior of the electro-acoustic transducer that emulates the transducer defined by said series of parameters set.

Visual information is used to alter or set an operating parameter of an audio signal processor, other than a beamformer. A digital camera captures visual information about a scene that includes a human speaker and / or a listener. The visual information is analyzed to ascertain information about acoustics of a room. A distance between the speaker and a microphone may be estimated, and this distance estimate may be used to adjust an overall gain of the system. Distances among, and locations of, the speaker, the listener, the microphone, a loudspeaker and / or a sound-reflecting surface may be estimated. These estimates may be used to estimate reverberations within the room and adjust aggressiveness of an anti-reverberation filter, based on an estimated ratio of direct to indirect (reverberated) sound energy expected to reach the microphone. In addition, orientation of the speaker or the listener, relative to the microphone or the loudspeaker, can also be estimated, and this estimate may be used to adjust frequency-dependent filter weights to compensate for uneven frequency propagation of acoustic signals from a mouth, or to a human ear, about a human head.

A mobile, wearable imaging system, having operator eyewear with a plurality of semi-transparent lenses, a microphone, a video capture device, and an accelerometer, wherein the semi-transparent lenses provide the operator a full-forward head's up view and a view of an external image superposed on the full-forward head's up view. The system also includes a mobile multifunction controller including an image processor that receives the external image and produces the view of an external image superposed on the full-forward head's up view in the operator eyewear, an audio signal processor, a motion signal processor, a gesture sensor, a manual input device, and a general processor unit, which executes a received system command, and where symbolic data received by the controller is selectively combined with the external image superposed on the full-forward head's up view.

A vehicle audio system includes at least two loudspeakers incorporated into a headrest of a vehicle. Protective caps are provided for the loudspeakers at the headrest above each of the two loudspeakers and extend at least partially in a direction in which sound is emitted from the loudspeakers. An audio signal processor receives an audio input signal and is configured for generating audio output signals for the two loudspeakers such that the audio output signals are perceived by a user sitting on a seat on which the headrest is provided as a virtual soundfield. A database includes cap compensating information that compensates for an influence by the protective caps on the audio output signals emitted by the two loudspeakers. The audio signal processor is further configured for generating the virtual soundfield for said user taking into account the cap compensating information.

An anti-noise earmuff device includes an audio signal processor unit, a Bluetooth module, an audio mixer unit and an amplifier unit. The audio signal processor unit is arranged in the earmuffs for control audio signal output, protecting the user's eardrums against noise of high-decibel level. The Bluetooth module receives or communicates with a mobile communication device through an antenna, allowing the user to listen to the music and to communicate with people outside without needing taking off the earmuff device.

A device for determining a component signal for a WFS system includes a provider for providing WFS parameters, a WFS parameter interpolator, and an audio signal processor. The provider provides WFS parameters for a component signal while using a source position and while using the loudspeaker position at a parameter sampling frequency smaller than the audio sampling frequency. The WFS parameter interpolator interpolates the WFS parameters so as to produce interpolated WFS parameters which are present at a parameter interpolation frequency that is higher than the parameter sampling frequency, the interpolated WFS parameters having interpolated fractions which have a higher level of accuracy than is specified by the audio sampling frequency. The audio signal processor is configured to apply the interpolated fractional values to the audio signal such that the component signal is obtained in a state of having been processed at the higher level of accuracy.

Visual information is used to alter or set an operating parameter of an audio signal processor, other than a beamformer. A digital camera captures visual information about a scene that includes a human speaker and / or a listener. The visual information is analyzed to ascertain information about acoustics of a room. A distance between the speaker and a microphone may be estimated, and this distance estimate may be used to adjust an overall gain of the system. Distances among, and locations of, the speaker, the listener, the microphone, a loudspeaker and / or a sound-reflecting surface may be estimated. These estimates may be used to estimate reverberations within the room and adjust aggressiveness of an anti-reverberation filter, based on an estimated ratio of direct to indirect (reverberated) sound energy expected to reach the microphone. In addition, orientation of the speaker or the listener, relative to the microphone or the loudspeaker, can also be estimated, and this estimate may be used to adjust frequency-dependent filter weights to compensate for uneven frequency propagation of acoustic signals from a mouth, or to a human ear, about a human head.

A television (10) for use in a home cinema or surround sound system is provided with an additional audio input (58) for a center audio channel signal from a multi-channel surround sound receiver / amplifier (38). Through system control logic, a control and switching stage (28) within the television couples (72) the televisions internal audio channel signal processing and amplifier stages (GTV,ATV) in parallel to the input (58) when in the surround sound mode, at the same time disabling the audio signal path from the televisions tuner (60) to the audio signal processors (GTV,ATV) and setting (74) the volume level on all channels to a predetermined level to allow volume control (63) from the remote source (38). A home cinema system featuring such a television configuration is also described.

An audio signal processing apparatus including a receiving unit receiving an input audio signal, a processor generating an output audio signal reproducing a virtual sound source corresponding to the input audio signal in a virtual space, and an output unit outputting an output audio signal generated by the processor is disclosed. The processor may obtain spatial information related to the virtual space including a virtual sound source corresponding to the input audio signal and a listener, filter the input audio signal based on a location of the virtual sound source and the spatial information to generate at least one reflected sound corresponding to each of at least one mirror plane in the virtual space, obtain a relative location of a virtual reflect sound source with respect to a location and a view-point of the listener, based on information of the view-point of the listener and a location of the virtual reflect sound source corresponding to each of the at least one reflected sound, and binaural render the at least one reflected sound, based on the relative location of the virtual reflect sound source corresponding to each of the at least one reflected sound.

A processing device includes a receiving unit to receive a program, a video signal processor to process a video signal of the program, and an audio signal processor to process an audio signal of the program. A remoter control or on screen display is used for inputting a user's command. A controller controls the video signal processor and the audio signal processor according to the user's command, so that the video signal and the audio signal are simultaneous adjusted. In an alternate embodiment, the controller automatically adjusts at least one of the video signal and the audio signal based upon a temperature in the environment of the device, genre data associated with the program, or an environment of a scene in the program, as determined by data associated with the program.

An audio signal processor forms gaps or guard bands in sequences of blocks conveying encoded audio information and time aligns the guard bands with video information. The guard bands are formed to allow for variations in processing or circuit delays so that the routing or switching of different streams of video information with embedded audio information does not result in a loss of any encoded audio blocks.

A digital A / V apparatus provides reduced channel changes times and maintains synchronization between audio and video content during a channel change event. According to an exemplary embodiment, the digital A / V apparatus includes at least one signal receiver for receiving a first audio stream, a first video stream and a second video stream, wherein the first and second video streams represent the same program and the first video stream has at least one characteristic different from the second video stream; a first audio signal processor for processing the first audio stream to generate a first processed audio stream; a first video signal processor for processing the first video stream to generate a first processed video stream; a second video signal processor for processing the second video stream to generate a second processed video stream having a delay with respect to the first processed video stream; a first buffer for buffering the first processed audio stream to provide de-jittering; a renderer for rendering the first processed audio stream and the second processed video stream in response to a channel change command; a switch for switching from the second processed video stream to the first processed video stream after the renderer begins rendering the second processed video stream, and thereby causing the renderer to begin rendering the first processed video stream; and whereby synchronization between the first processed audio stream and the first processed video stream is maintained during a channel change event.

An audio network system that performs transport of audio signals among nodes by cascading a plurality of nodes each including two sets of transmission I / Fs and reception I / Fs, and circulating among the nodes in each fixed period an audio transport frame generated by a master node, the audio transport frame including a plurality of storage regions for audio signals, is configured such that the master node measures time periods Dfw and Dbw after the audio transport frame is transmitted until the audio transport frame returns to the master node after passing through the transmission route and writes them into the audio transport frame, and each of the other nodes generates a signal processing wordclock based on those time periods, two reception times Tr1 and Tr2 while the audio transport frame circulates once through the transmission route, and a predetermined target time Tt.

Embodiments include methods and apparatuses for echo cancelation involving multiple audio channels and the production, sensing, or a combination thereof of multiple audio channels in conferencing systems. A conferencing apparatus with a plurality of speakers configured to generate outgoing acoustic waves responsive to a multi-channel audio signal. One or more microphones are configured to sense incoming acoustic waves from the plurality of speakers and from locally produced acoustic waves from a participant of a conference to generate one or more incoming audio signals. A processor is operably coupled to the plurality of speakers and the one or more microphones. The processor is configured to perform acoustic echo cancelation on the one or more incoming audio signals relative to at least two different channels of the multi-channel audio signal.

An apparatus for processing an audio signal includes a configurable first audio signal processor for processing the audio signal in accordance with different configuration settings to obtain a processed audio signal, wherein the apparatus is adapted so that different configuration settings result in different sampling rates of the processed audio signal. The apparatus furthermore includes an analysis filter bank having a first number of analysis filter bank channels, a synthesis filter bank having a second number of synthesis filter bank channels, a second audio processor being adapted to receive and process an audio signal having a predetermined sampling rate, and a controller for controlling the first number of analysis filter bank channels or the second number of synthesis filter bank channels in accordance with a configuration setting.

In the case of forming an audio network system in which a plurality of processors each having two sets of transmission I / Fs and reception I / Fs are connected and a TL frame generated by a master node and including a plurality of storage regions for audio signals circulates among the processors in each constant period to transport audio signals the among processors, the system is configured such that when a transmitted TL frame does not return to the reception I / F on the forward side but returns to the reception I / F on the backward side in a single mode of transmitting the TL frame to the forward side and circulating the TL frame through one transmission route, the system shifts to a twin mode of transmitting and circulating TL frames through two transmission routes.

An audio network system that performs transport of audio signals between nodes by cascading a plurality of nodes each including two sets of transmission I / Fs and reception I / Fs, and circulating among the nodes in each fixed period an audio transport frame generated by a master node, the audio transport frame including a plurality of storage regions for audio signals, is configured such that when the configuration of the system is changed, loopback of the audio transport frame in each node and release thereof are automatically controlled to automatically loop back the transport of the audio transport frame at nodes at both ends and not loop back it at the other nodes to thereby form a loop transmission route among the nodes.

An audio signal processor receives a plurality of encoded multi-channel audio signals. A multi-channel decoder (105) decodes a first encoded multi-channel signal to generate a first decoded multi-channel signal. A generator (109) generates an encoded further audio signal by selecting audio encoding data from at least a second encoded multi-channel audio signal such that a number of channels of the encoded further audio signal comprising audio encoding data from the second encoded multi-channel audio signal is less than a number of channels in the second encoded multi-channel signal. Thus, a channel reduction is performed in the encoded data domain. A further decoder (111) generates a further decoded signal by decoding the further encoded audio signal. A combiner (107) combines the first decoded multi-channel signal and the further decoded signal to generate a multi-channel output signal. An exciting user experience can be provided while maintaining low complexity and resource usage.

A hearing prosthesis arrangement is described for a hearing assisted patient. A microphone senses an acoustic environment around the hearing assisted patient and generates a corresponding microphone output signal. An audio signal processor processes the microphone output signal and produces a corresponding prosthesis stimulation signal to the patient for audio perception. The audio signal processor includes a dereverberation process that measures a dedicated reverberation reference signal produced in the acoustic environment to determine reverberation characteristics of the acoustic environment, and reduces reverberation effects in the hearing prosthesis stimulation signal based on the reverberation characteristics.

A portable terminal includes an apparatus capable of providing a hearing aid service. The apparatus includes a sensor configured to measure a spatial tilt of the portable terminal. The apparatus also includes a controller that determines whether a left ear or a right ear is used for communication based on the measured spatial tilt of the portable terminal. An audio signal processor compensates an audio signal according to the characteristic of determined ear.

The invention provides an audio signal processing device capable of improving sound quality by causing a voice switch to operate appropriately. Delay-subtraction processing is performed on an input signal to form a first and second directional signal with nulls in a first and second specific direction, respectively, and a coherence is obtained using the two directional signals. The coherence is then compared to a determination threshold value to determine whether the input audio signal is a target-sound segment arriving from a target-direction, or a non-target-sound segment other than the target-sound segment. A gain is set according to the determination result, and any non-target-sound is attenuated by multiplying the input signal by the gain. The determination threshold value is controlled based on an average value of coherence in interfering-sound segments.

Method and apparatus for generating a composite audio signal clocked to a selected frequency from left and audio input signals. A video / audio signal generator transmits the left and right audio input signals, together with a video signal having a horizontal synchronization signal embedded therein. If copy protection has not been applied to the video signal, the embedded horizontal synchronization signal will have a frequency of 15.734 KHz. If applied to the video signal, copy protection techniques will shift the frequency of the horizontal synchronization signal. To avoid distortion of the composite audio signal, a switch configured to selectively couple the video signal or a secondary video signal generated by an internal video signal generator to an audio signal processor which generates the composite audio output signal from the left and right audio input signals. Embedded within the secondary video signal is a horizontal synchronization signal having a frequency of 15.734 KHz. Thus, if copy protection features cause the frequency of the signal embedded within the primary video signal to shift, the switch may be used to couple the audio signal processor to the internal video signal generator, thereby ensuring that the composite audio signal will always be clocked to the selected frequency.

In the case where an audio network system that performs transport of audio signals between a plurality of processors is constructed such that the plurality of processors each including two sets of transmission I / Fs and reception I / Fs are connected in series, and a TL frame generated by a master node is circulated along a ring transmission route formed among the processors in each fixed period, the TL frame including a plurality of storage regions for audio signals, when one of the processors is required to be the master node, the one processor resets in sequence processors which are connected on both sides thereof and processors behind the processors connected on both sides to once remove each of those processors from a ring transmission route if it belong to any ring transmission route, then incorporates the processors into a ring transmission route in which the one processor belongs to.

Audio streaming is made available throughout the signal processing path of the speech processor of a cochlear implant or other audio signal processor. Audio streaming comprises the digitally phase locked playback of a real time n-bit digital audio stream, where n may be a large number, e.g., 8, 12, 16, 24 or 32, that emanates (unsolicited) from an operating speech processor. A number of sample points are made available long the processing chain of a digital signal processor (DSP) used within the speech processor of the cochlear implant. Audio streaming may occur at any sample point. The signal at a selected sample point may be selectively monitored in order to allow appropriate diagnostics to be performed. Audio streaming utilizes an auto-referencing mixed-mode phase locked loop. Such phase locked loop processes an asynchronous stream of digital audio samples that arrive at a designated location, e.g., a selected sample point, at a consistent, but unknown, average rate. Once the stream of asynchronous audio samples are received, the average sample rate is extracted, and a local clock is generated at the average frequency. The incoming audio streaming samples are then re-synchronized with the local clock. in order to allow digitally phase locked playback of the audio stream.

An audio network system that performs transport of audio signals among nodes by cascading a plurality of nodes each including two sets of transmission I / Fs and reception I / Fs, and circulating among the nodes in each fixed period an audio transport frame generated by a master node, is configured such that the master node generates the audio transport frame in an S-th period based on the audio transport frame in an (S−k)-th period, and each of the other nodes delays the audio signals written by another node after the audio transport frame is generated by the master node until the audio transport frame is transmitted to the self node, by k period(s) with respect to the other audio signals, for use in signal processing.

The invention provides a method for collecting and playing audio signals in real time, a mobile terminal and a system. The mobile terminal comprises an audio signal transceiver which is matched and connected with at least one audio collector outside the mobile terminal to receive audio signals collected by the matched and connected audio collector and transmit the received audio signals to an audio signal processor; and the audio signal processor which receives the audio signals sent by the audio signal transceiver, and outputs and plays the received audio signal mixed with the audio signals corresponding to the background sounds stored in a database. The recording and play effects of the audio signals can be improved. The mobile terminal can be connected with at least one audio collector so that audio shared by persons can be collected and played.

The invention regards a scheme for generating a probe noise signal to be used in an anti feedback system of an audio system. The audio system comprises e.g. a microphone for capturing an audio signal, an audio signal processor for adaptation of the audio signal and a receiver for generation of an audible signal. According to an embodiment of the invention, a noise signal is injected into the audio signal path between the microphone and the receiver and used for estimating acoustical feedback, the noise signal being generated by the following steps:converting a digitized audio signal to the frequency domain, in order to obtain a series of magnitude and phase values,changing the phase values such that the phase of the resulting signal becomes less correlated, preferably substantially un-correlated, to the original signal,converting the magnitude and phase back to a time domain signal using the changed phase values.The invention may e.g. be used in a hearing aid, a headset or a pair of headphones.