Abstract: The present technology relates to signal processing device and method that make it possible to reproduce sound more effectively. A signal processing device includes a rotation operation unit that rotates a head-related transfer function in a spherical harmonic domain by an operation on the basis of a rotation matrix corresponding to rotation of a head of a listener, the operation in which an order of the rotation matrix is limited, and a synthesis unit that synthesizes the head-related transfer function after rotation obtained by the operation and a sound signal of the spherical harmonic domain to generate a headphone drive signal. The present technology is applicable to an audio processor.

Abstract: A mobile computing device stand with a holder for a mobile computing device with a media player for conducting an audio session and back speakers positioned in the seat behind a location where a user is expected to be positioned and next to the seat. The back speakers include a first set positioned at a first side of the seat and a second set positioned at a second side of the seat opposite the first side of the seat. The first and second sets includes first and second inward back speakers facing toward the location and first and second outward back speakers facing away from the location. A system controller receives audio signals with audio of an audio session, sends the audio signals to the first and second inward back speakers, and sends the audio signals in an out of phase manner to the first and second outward back speakers.

Abstract: The present technology relates to a decoding apparatus, a decoding method and a program which make it possible to obtain sound with higher quality. A demultiplexing circuit demultiplexes an input code string into a gain code string and a signal code string. A signal decoding circuit decodes the signal code string to output a time series signal. A gain decoding circuit decodes the gain code string. That is, the gain decoding circuit reads out gain values and gain inclination values at predetermined gain sample positions of the time series signal and interpolation mode information. An interpolation processing unit obtains a gain value at each sample position between two gain sample positions through linear interpolation or non-linear interpolation according to the interpolation mode based on the gain values and the gain inclination values. A gain applying circuit adjusts a gain of the time series signal based on the gain values. The present technology can be applied to a decoding apparatus.

Abstract: An apparatus is disclosed for current-mode filtering using current steering. In an example aspect, the apparatus includes a filter. The filter includes a current-steering node, a first output node, a second output node, a wideband path, and a narrowband path. The wideband path is coupled between the current-steering node and the first output node. The wideband path includes a wideband low-pass filter configured to pass frequencies within a wide passband. The narrowband path is coupled between the current-steering node and the second output node. The narrowband path includes a narrowband low-pass filter configured to pass a portion of the frequencies that are within a narrow passband.

Abstract: An apparatus for decoding an encoded audio signal to obtain modified output signals includes an input interface for receiving a transmitted downmix signal and parametric data relating to audio objects included in the transmitted downmix signal, the downmix signal being different from an encoder downmix signal, to which the parametric data is related; a downmix modifier for modifying the transmitted downmix signal using a downmix modification function, wherein the downmix modification is performed in such a way that a modified downmix signal is identical to the encoder downmix signal or is more similar to the encoder downmix signal compared to the transmitted downmix signal; an object renderer for rendering the audio objects using the modified downmix signal and the parametric data to obtain output signals; and an output signal modifier for modifying the output signals using an output signal modification function.

Abstract: An audio processing circuit is described. The audio processing circuit includes at least one input configured to receive a primary audio signal and a feature generation signal. A feature model estimation circuit is configured to model and output a feature model signal of the primary audio signal. A feature generation circuit is coupled to the feature model estimation circuit and is to receive the feature model signal and the feature generation signal and, in response to the feature model signal, modify the feature generation signal; and output a modified representation of the feature generation signal that is more similar to the primary audio signal.

Abstract: Example audio signal processing apparatuses and methods for processing an input audio signal into an output audio signal are disclosed. An example audio signal processing apparatus comprises a decomposer configured to decompose the input audio signal into a direct audio signal and a diffuse audio signal, a modifier configured to modify the direct audio signal in order to obtain a modified direct audio signal, wherein the modifier comprises a bandwidth extender configured to extend an upper cutoff frequency of a frequency range of the direct audio signal, and a combiner configured to combine the modified direct audio signal with the diffuse audio signal to obtain the output audio signal.

Abstract: A decoder operable to decode audio signals. The decoder operable to receive an encoded bitstream that includes bitstream synchronization command data and program command data and process the encoded bitstream and identify within the bitstream the synchronization command data. The decoder further operable to decode the program command packet and at least one program related channel data using information provided in the synchronization command data and decode program related channel data using information provided in the program command data.

Abstract: A phone stand includes a phone holder for coupling to a phone, where the phone is for conducting a voice session. The phone stand further includes a plurality of directional speakers positioned to project sound to a focused audio area corresponding to a location where a user is expected to be positioned, a plurality of back speakers positioned behind the location where the user is expected to be positioned, and a system controller. The system controller is configured to: receive audio signals, where the audio signals includes audio of the voice session from the phone; separate the audio signals into speech signals and non-speech signals; obtain one or more output mixing attributes; generate mixed signals by combining the speech signals and the non-speech signals according to the one or more output mixing attributes; and send the mixed signals to the plurality of directional speakers.

Abstract: A method for processing an FM stereo signal. The FM stereo signal is digitized and divided into overlapping blocks, which are transformed into the frequency domain. Individual spectral lines of the difference signal are lowered if these have a higher magnitude than the respective spectral lines of the sum signal. The sum and difference signals are then transformed back.

Abstract: A phase compensation circuit includes: a first circuit that increases phase characteristic of a specific frequency of an electrical signal; a second circuit that decreases the phase characteristic of the specific frequency of the electrical signal; and a limiting amplifier that amplifies an electrical signal that is processed by at least one of the first circuit and the second circuit.

Abstract: Aspects of the subject disclosure may include, for example, embodiments receiving audio content in a multi-channel sound format over a communication network resulting in multi-channel audio content. Further embodiments can include identifying a compression ratio of the audio content. Additional embodiments can include determining a rendered sound externalization for rendering the audio content according to the compression ratio of the audio content. Also, embodiments can include rendering the audio content in a binaural audio format for headphone playback on an audio device according to the rendered sound externalization. Other embodiments are disclosed.

Abstract: According to one embodiment, a method performed by an electronic device includes: receiving an audio signal comprising voice and background sound via a microphone; receiving a user's operation to set a loudness of the voice or the background sound; setting a balance between a first gain of the voice and a second gain of the background sound according to the user's operation; separating the input audio signal into a first signal of the voice and a second signal of the background sound; amplifying the first signal according to the first gain; amplifying the second signal according to the second gain; and outputting the first signal and the second signal at least partially overlapping each other via a speaker.

Abstract: Embodiments of the present invention relate to audio object extraction. A method for audio object extraction from audio content of a format based on a plurality of channels is disclosed. The method comprises applying audio object extraction on individual frames of the audio content at least partially based on frequency spectral similarities among the plurality of channels. The method further comprises performing audio object composition across the frames of the audio content, based on the audio object extraction on the individual frames, to generate a track of at least one audio object. Corresponding system and computer program product are also disclosed.

Abstract: A system for processing audio data of the present disclosure has an audio processing device for receiving audio data from an audio source. Additionally, the system has logic that separates the audio data received into left channel audio data indicative of sound from a left audio source and right channel audio data indicative of sound from a right audio source. The logic further separates the left channel audio data into primary left ear audio data and opposing right ear audio data and for separating the right channel audio data into primary right ear audio data and opposing left ear audio data applies a first filter to the primary left ear audio data, a second filer to the opposing right ear audio data, a third filter to the opposing left ear audio data, and a fourth filter to the primary right ear audio data, wherein the second and third filters introduce a delay into the opposing right ear audio data and the opposing left ear audio data, respectively.

Abstract: A method, medium, and system decoding and/or encoding multiple channels. Accordingly, down-mixed multiple channels can be decoded/up-mixed to a left channel and a right channel during a first stage, thereby enabling a high quality sound output even in scalable channel decoding.

Abstract: There is provided an audio signal processing device for setting, upon outputting audio signals from a left speaker and a right speaker each having an identical amplitude-frequency characteristic and a phase characteristic, and from one center speaker arranged between the left speaker and the right speaker, listening points between the left speaker and the center speaker and between the right speaker and the center speaker, the audio signal processing device including an inverse filter processing unit configured to provide inverse characteristics of an amplitude-frequency characteristic and a phase characteristic of the center speaker, and a speaker characteristic adding processing unit configured to correct the amplitude-frequency characteristic and the phase characteristic of an audio signal of the center speaker that have been processed by the inverse filter processing unit in a manner that they correspond to characteristics of the left speaker and the right speaker.

Abstract: An apparatus for spatial sound reproduction comprises a receiver (101) for receiving a multi-channel audio signal. An analyzer (107) determines a spatial property of the multi-channel audio signal, such as a spatial complexity or organization. A selection processor (109) then selects a reproduction mode from a plurality of sound reproduction modes where the multi-channel sound reproduction modes employ different spatial rendering techniques. A reproduction circuit (103) then drives a set of loudspeakers (105) to reproduce the multi-channel audio signal using the selected reproduction mode. The switching between the reproduction modes may be fast (e.g. in the order of 100 ms to 10 secs) thereby allowing a short term adaptation of the reproduction mode to the signal characteristics. The approach may in particular provide an improved spatial experience to a listener.

Abstract: Wireless microphone systems are used, for example, in theater, stage performances, television, broadcast radio or in concerts to amplify voices or instruments for a large audience. Due to the plurality of actors or musicians, a corresponding number of audio channels is required to transmit the audio information. 20 to 50 audio channels can thus be required in a parallel mode even in common applications.

Abstract: A method of optimizing stereo reception for an analog radio by applying the demodulated right sound signal (SD) and left sound signal (SG) as input to a decorrelation module having a variable decorrelation rate. The decorrelation rate of the decorrelation module is modified as a function of the reception quality coefficient “alpha” provided by the radio. The decorrelation module applies a higher decorrelation rate for a smaller reception quality coefficient “alpha” and applies a lower decorrelation rate for a larger reception quality coefficient “alpha. Also, a module for generating high-pitched sounds to recreate the high-frequency component (SHF) of the right or left sound signals which has been removed in the event of poor reception.

Abstract: The invention relates to an apparatus for improving a stereo audio signal of an FM stereo radio receiver. The apparatus comprises a parametric stereo (PS) parameter estimation stage. The parameter estimation stage is configured to determine 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 apparatus comprises an upmix stage. The upmix stage is configured to generate 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 in a downmix stage. The PS parameter estimation stage may be part of a PS encoder. The upmix stage may be part of a PS decoder.

Abstract: An audio receiver may include a mono/stereo detector that causes the audio receiver to output either a monophonic or a stereophonic signal based on a difference between a pilot energy signal and a filtered pilot energy signal. The audio receiver includes a filter that filters the pilot energy signal to generate a filtered pilot energy signal, wherein a variable leakage factor associated with the filter is used to minimize a noise level of the filtered pilot energy signal and to reduce a response time of the audio receiver.

Abstract: The present technology relates generally to a closed-loop power stage and audio amplifier circuits comprising the same. The present technology further provides a method for controlling an amplifier circuit. The closed-loop power stage is configured to receive a PWM input signal having a first frequency and a first duty cycle, a power supply voltage, and a bias signal related to the power supply voltage, and to output a square-wave signal having a second frequency and a second duty cycle. The closed-loop power stage comprises a feedback loop which is configured to regulate the second duty cycle of the square-wave signal in response to a variation in the power supply voltage. The audio amplifier circuit comprising the closed-loop power stage may have a stable output voltage and thus have good power supply rejection performance.

Abstract: A method for suppressing ambient noise using multiple audio signals may include providing at least two audio signals captured by at least two electro-acoustic transducers. The at least two audio signals may include desired audio and ambient noise. The method may also include performing beamforming on the at least two audio signals in order to obtain a desired audio reference signal that is separate from a noise reference signal.

Abstract: An apparatus for encoding/decoding a multichannel signal. The apparatus for encoding/decoding a multichannel signal processes phase parameters for phase information among a plurality of channels constituting the multichannel signal in consideration of the characteristics of the multichannel signal. The apparatus generates an encoded bit stream for the multichannel signal using the processed phase parameters and the mono signal extracted from the multichannel signal.

Abstract: Various methods and devices are provided for a wireless audio system for a number of users. The system includes a base unit that is adapted to removably store, recharge and communicate with various communication modules, including personal microphone modules, table-top microphones, and audio adapters. The system also includes a plurality of personal microphone modules that are each adapted to be removable and coupled, for example, to a user's clothing, and further, are adapted to communicate wirelessly with the base unit, and table-top microphones that are adapted to communicate wirelessly with the base unit.

Abstract: A passive, non-amplified audio switch for a computer telephony system is provided. The passive, non-amplified audio switch provides a first telephone jack to receive audio in and send audio out to a first user. The passive, non-amplified audio switch provides a second telephone jack to receive audio in and send audio out to a second user. The passive, non-amplified audio switch further provides audio out and receives audio in from a voice platform, which transcribes audio using a speech to text engine. The passive, non-amplified audio switch provides that audio in from the second user is always provided to the voice platform for transcription.

Abstract: In order to generate a multi-channel signal having a number of output channels greater than a number of input channels, a mixer is used for upmixing the input signal to form at least a direct channel signal and at least an ambience channel signal. A speech detector is provided for detecting a section of the input signal, the direct channel signal or the ambience channel signal in which speech portions occur. Based on this detection, a signal modifier modifies the input signal or the ambience channel signal in order to attenuate speech portions in the ambience channel signal, whereas such speech portions in the direct channel signal are attenuated to a lesser extent or not at all. A loudspeaker signal outputter then maps the direct channel signals and the ambience channel signals to loudspeaker signals which are associated to a defined reproduction scheme, such as, for example, a 5.1 scheme.

Abstract: A method of processing a signal, including taking a signal formed from a plurality of source signal emitters and expressed in an original domain, decomposing the signal into a mathematical representation of a plurality of constituent elements in an alternate domain, analyzing the plurality of constituent elements to associate at least a subset of the constituent elements with at least one of the plurality of source signal emitters, separating at least a subset of the constituent elements based on the association and reconstituting at least a subset of constituent elements to produce an output signal in at least one of the original domain, the alternate domain and another domain.

Abstract: Provided are a method and apparatus for encoding and decoding a stereo signal or a multi-channel signal. According to the method and apparatus, a stereo signal or a multi-channel signal can be encoded and/or decoded by generating parameters based on a mono signal.

Abstract: An apparatus for encoding/decoding a multichannel signal. The apparatus for encoding/decoding a multichannel signal processes phase parameters for phase information among a plurality of channels constituting the multichannel signal in consideration of the characteristics of the multichannel signal. The apparatus generates an encoded bit stream for the multichannel signal using the processed phase parameters and the mono signal extracted from the multichannel signal.

Abstract: A sound processing apparatus is provided for estimating the power of background noise using a directional sound receiving technology using a plurality of sound receiving units, computing a gain control value on the basis of the estimated power of background noise and a predetermined power target value, and outputting the gain control value, so that a delay time of starting gain control can be reduced, and a slow response of a speech recognition application program or degradation of the speech quality of a voice communication program can be prevented.

Abstract: A method for processing an audio signal, comprising: receiving a downmix signal, a first multi-channel information, and an object information; processing the downmix signal using the object information and a mix information; and, transmitting one of the first multi-channel information and a second multi-channel information according to the mix information, wherein the second channel information is generated using the object information and the mix information is disclosed.

Abstract: Widening stereophonic response is achieved in sound reproduction systems with at least two loudspeakers. A stereo signal input is accessed, which includes multiple frequency components. The loudspeakers are close to each other. A frequency range of the frequency components is decorrelated, e.g., upon pre-processing the stereo signal. The sound reproduction system's stereophonic response is widened, based on the decorrelation.

Abstract: A device for processing an audio signal, wherein the device comprises a detection unit adapted for detecting a widening state of the audio signal, and a widening modification unit adapted for modifying a widening characteristic of the audio signal depending on the detected widening state of the audio signal.

Abstract: An FM audio receiver can include a mono/stereo detector that causes the audio receiver to output either a monophonic or a stereophonic signal based on a received pilot tone energy. An accurate operation of the receiver, including but not limited to correct decoding of monophonic/stereophonic reception, can be based on the receiver operating with the same rated maximum system deviation (RMSD) as the received signal itself. Aspects of the disclosure describe a system and method of detecting and matching a receiver's RMSD to that of a received signal by demodulating a carrier bearing a an input signal over a first bandwidth, extracting a pilot energy signal from the input signal, and demodulating the carrier bearing the input signal over a second bandwidth if the pilot energy signal is within a pilot energy range for a first predetermined amount of time.

Abstract: For generating a parametric representation of a multi-channel signal especially suitable for low-bit rate applications, only the location of the maximum of the sound energy within a replay setup is encoded and transmitted using direction parameter information. For multi-channel reconstruction, the energy distribution of the output channels identified by the direction parameter information is controlled by the direction parameter information, while the energy distribution in the remaining ambience channels is not controlled by the direction parameter information.

Abstract: Systems provide an audio/visual output to each of a plurality of listeners in a manner that permits individualized audio adjustment, wherein audio comprises a first signal that is substantially voice and a second signal that is substantially other than voice. The systems may include a video device, a storage medium, and a transmitter that transmits the first and second signals to a plurality of personal listening devices. Each of the plurality of personal listening devices may include first and second receivers, first and second adjustment devices, an audio signal combining device, and one or more transducers, wherein the systems permit each of the plurality of listeners to adjust the first and second signals independently of other ones of the plurality of listeners in an audience.

Abstract: A wireless listening system includes a base operable to receive an electrical signal corresponding to an audio signal, including the ring of an incoming telephone call, up-convert the audio signal to an infrared signal, and transmit the infrared audio signal wirelessly therefrom, and a wireless headset operable to selectably receive either the infrared audio signal transmitted by the base, or an infrared signal corresponding to an audio signal transmitted by a public address system, down-convert the received signal back into the audio signal, and audibly reproduce the audio signal to a wearer of the headset. The system enables a hearing impaired listeners to enjoy their favorite audio programs without disturbing nearby persons with normal hearing and without missing an incoming telephone call, and is also compatible with public IR PA systems currently used to assist the hearing impaired community.

Abstract: A method includes splitting and filtering a left input signal and a right input signal to produce a plurality of frequency sub-bands. Each of the frequency sub-bands includes a left sub-band signal and a right sub-band signal. The method also includes processing the left and right sub-band signals associated with each of the frequency sub-bands into a plurality of sub-band channel signals. The plurality of sub-band channel signals includes at least three sub-band channel signals. In addition, the method includes summing corresponding ones of the sub-band channel signals for reproduction in a corresponding channel of a plurality of channels. The plurality of sub-band channel signals may include a left sub-band channel signal, a right sub-band channel signal, a center sub-band channel signal, a left surround sub-band channel signal, and a right surround sub-band channel signal.

Abstract: The present invention is based on the finding that a reconstructed output channel, reconstructed with a multi-channel reconstructor using at least one downmix channel derived by downmixing a plurality of original channels and using a parameter representation including additional information on a temporal fine structure of an original channel can be reconstructed efficiently with high quality, when a generator for generating a direct signal component and a diffuse signal component based on the downmix channel is used. The quality can be essentially enhanced, if only the direct signal component is modified such that the temporal fine structure of the reconstructed output channel is fitting a desired temporal fine structure, indicated by the additional information on the temporal fine structure transmitted.

Abstract: Provided are a method and apparatus for encoding and decoding a stereo signal or a multi-channel signal. According to the method and apparatus, a stereo signal or a multi-channel signal can be encoded and/or decoded by generating parameters based on a mono signal.

Abstract: A sound source signal from a target sound source is allowed to be separated from a mixed sound which consists of sound source signals emitted from a plurality of sound sources without being affected by uneven sensitivity of microphone elements. A beamformer section 3 of a source separation device 1 performs beamforming processing for attenuating sound source signals arriving from directions symmetrical with respect to a perpendicular line to a straight line connecting two microphones 10 and 11 respectively by multiplying output signals from the microphones 10 and 11 after spectrum analysis by weighted coefficients which are complex conjugate to each other. Power computation sections 40 and 41 compute power spectrum information, and target sound spectrum extraction sections 50 and 51 extract spectrum information of a target sound source based on a difference between the power spectrum information.

Abstract: The invention enables the inclusion of voice and remaining audio information at different parts of the audio production process. In particular, the invention embodies special techniques for VRA-capable digital mastering, accommodation of PCPV/PCA and/or SCRA signals in audio CODECs, VRA-capable encoders and decoders, and VRA in DVD and other digital audio file formats. The invention facilitates an end-listener's voice-to-remaining audio (VRA) adjustment upon the playback of digital audio media formats by focusing on new configurations of multiple parts of the entire digital audio system, thereby enabling a new technique intended to benefit audio end-users (end-listeners) who wish to control the ratio of the primary vocal/dialog content of an audio program relative to the remaining portion of the audio content in that program.

Abstract: An audio signal is processed to derive primary and ambient components of the signal. The signal is first transformed to generate frequency-domain subband signals. Primary and ambient components are separated by comparing frequency subband content using a complex-valued similarity metric, wherein one of the primary and ambient components is determined to be the residual after the other is identified using the similarity metric.

Abstract: An audio receiver may include a mono/stereo detector that causes the audio receiver to output either a monophonic or a stereophonic signal based on a difference between a pilot energy signal and a filtered pilot energy signal. The audio receiver includes a filter that filters the pilot energy signal to generate a filtered pilot energy signal, wherein a variable leakage factor associated with the filter is used to minimize a noise level of the filtered pilot energy signal and to reduce a response time of the audio receiver.

Abstract: In a situation where a plurality of virtual sound sources are fixedly located, a mixing system allows a listener to increase or decrease the sound volumes of the virtual sound sources with a simple operation. Directional angle difference detection means 24 detects a directional angle difference between a predetermined reference directional angle and an indicated directional angle which is determined by the listener by pointing with an operation unit 7 a certain direction in a virtual space where the virtual sound sources exist. Mixing means 5 mixes a plurality of audio signals supplied from a plurality of audio channels and outputs a mixed audio signal without adjustment, when the directional angle difference is zero degrees.

Abstract: The sound reproduction of a multichannel sound reproduction system with a plurality of speakers which is connected to the output of an FM stereo receiver is controlled by a control signal derived from the reception quality. Preferably, the control signal from the FM stereo receiver for controlling the stereo and mono components is also employed to control the multichannel sound reproduction system. For example, stereo, pseudo-stereo or mono reproduction are provided in the multichannel sound reproduction system in response to the stereo component within the output signal.

Abstract: A method for processing sound signal including an L side enhanced difference signal comprising a sum of a signal being removed low frequency component from L?R signal and an L+R signal, an R side enhanced difference signal comprising a sum of a signal being removed low frequency component from R?L signal and an R+L signal, an L side output signal comprising a subtraction of signals, in which said R side enhanced difference signal being delayed to remove low and high ranges and amplified by a predetermined amplifying rate for a purpose of cross cancellation talk is subtracted from said L side enhanced difference signal and an R side output signal comprising a subtraction of signals, in which said L side enhanced difference signal being delayed to remove low and high ranges and amplified by a predetermined amplifying rate for a purpose of cross cancellation talk is subtracted from said R side enhanced difference signal.

Abstract: An output reducing system for a telephone system, e.g., implemented in a headset, includes an input line and an output line, wherein the input line transmits an incoming signal to an output transducer and the output line transmits an outgoing signal from an input transducer, and a compressor system is provided in the input line for reducing the output signal level.