Abstract: In accordance with embodiments of the present disclosure, a method for operating a playback path comprising a first dynamic range enhancement subsystem and a second dynamic range enhancement subsystem, wherein an audio signal generated by the first dynamic range enhancement subsystem is communicated to the second dynamic range enhancement subsystem, is provided.

Abstract: According to a preferred aspect, there is provided a system and method for controlled dynamics adaptation of audio works. The dynamics complexity of the audio is adapted depending on the listening environment and its dynamics properties. The instant invention utilizes a function of stepwise linear and logarithmical sections, which describe individual points of a curve and which therefore with the use of a parameter can be utilized to calculate and control the target value of the loudness range. The curve points are stored and provided as metadata of the musical program, which allows the dynamics scaling functionality to be implemented dynamically and not tied to a certain time or place.

Abstract: A communication system includes a processor and a computer readable medium, coupled with the processor, comprising instructions that program the processor to: cause an electromagnetic emitter to irradiate a location on a selected object in proximity to a speaker; determine, from radiation reflected by the selected object at the irradiated location, an audio characteristic of the irradiated location; and when the audio characteristic is acceptable, assigning an electromagnetic microphone to collect audio information from radiation reflected by the selected object at the irradiated location.

Abstract: An organic light emitting display device includes a timing controller, data driver, and a plurality of pixels. The timing controller outputs gamma voltages corresponding to image data. The data driver generates data signals based on the gamma voltages. The pixels emit light with brightness corresponding to the data signals. The timing controller includes an on-pixel ratio (OPR) operator to calculate an OPR based on the image data and a gamma voltage supplier to select one of a plurality of gamma tables based on the OPR and to output gamma voltages stored in the selected one of the gamma tables.

Abstract: Disclosed is a signal processor for headphone off-ear detection. The signal processor includes an audio output to transmit an audio signal toward a headphone speaker in a headphone cup. The signal processor also includes a feedback (FB) microphone input to receive a FB signal from a FB microphone in the headphone cup. The signal processor also includes an off-ear detection (OED) signal processor to determine an audio frequency response of the FB signal over an OED frame as a received frequency response. The OED processor also determines an audio frequency response of the audio signal times an off-ear transfer function between the headphone speaker and the FB microphone as an ideal off-ear response. A difference metric si generated comparing the received frequency response to the ideal off-ear frequency response. The difference metric is employed to detect when the headphone cup is disengaged from an ear.

Abstract: A method of processing an audio signal is provided. The method includes receiving at least one digital audio stream from at least one audio source, applying at least one gain value to each of the at least one digital audio stream by a gain adjustment unit, mixing the at least one digital audio stream to which the at least one gain value has been applied, calculating a representative gain value from the at least one gain value, calculating an extra valid headroom based on the representative gain value for correction of the mixed at least one digital audio stream, and post-processing the mixed at least one digital audio stream using the extra valid headroom.

Abstract: Systems for controlling parameters of a MEMS microphone. In one embodiment, the microphone system includes a MEMS microphone and a controller. The MEMS microphone includes a movable electrode, a stationary electrode, and a driven electrode. The movable electrode is configured such that acoustic pressure acting on the movable electrode causes movement of the movable electrode. The stationary electrode and the driven electrode are positioned on a first side of the movable electrode. The driven electrode is configured to alter a parameter of the MEMS microphone based on a control signal. The controller is coupled to the stationary electrode and the driven electrode. The controller is configured to determine a voltage difference between the movable electrode and the stationary electrode. The controller is also configured to generate the control signal based on the voltage difference.

Abstract: According to a first aspect of the embodiments, a microphone mixer is provided comprising: an input adapted to receive differential microphone (mic) output signals; a gain-trim circuit adapted to receive the differential mic output signals, and which includes a substantially fully differential amplifier adapted to amplify the received differential mic output signals through use of a gain-trim output adjustment device that provides a variable gain amount ranging from a first gain-trim gain value to a second gain-trim gain value, to produce differential gain-trim circuit output signals; a fader circuit adapted to receive the differential gain-trim circuit output signals, and which includes a differential amplifier adapted to attenuate the received differential gain-trim circuit output signals through use of a fader output adjustment device that provides a variable gain amount ranging from a first fader gain value to a second fader value; and a common adjustment apparatus that mechanically ties the gain-trim out

Abstract: An electro-optical modulator is provided. The electro-optical modulator comprises at least one optical waveguide; a plurality of segments, wherein each one of the segments comprises at least one waveguide electrode for supplying a voltage to the optical waveguide, wherein the segments are arranged one behind the other along the optical waveguide, and wherein each one of the segments comprises its own driver unit electrically connected to the waveguide electrode of the segment for supplying an electric signal. Each one of the segments forms an electrical resonant circuit, wherein the segments are configured in such a way that the resonance frequency of the electrical resonant circuit of at least one of the segments lies within the frequency range of an electrical signal supplied to the driver units.

Abstract: A mixer includes a plurality of channel modules, and audio signals having processed in the individual channel modules are output from a main output. A temporary channel module is created and made usable in response to a preview start instruction for previewing a desired one of the channel modules as a preview-object channel module. The same audio signal as input to the preview-object channel module is input to the temporary channel module. The temporary channel module performs signal processing on the input audio signal independently of the preview-object channel module, and the audio signal having been subjected to the signal processing is output to a monitor output. Thus, a user is allowed to change a signal processing parameter value to be applied to an audio signal and preview the changed result, without influencing the main output.

Abstract: Systems, methods, apparatus, and articles of manufacture to facilitate playback of multimedia content are disclosed. An example apparatus includes a network interface configured to receive first audio content and second audio content over a playback network. A processor comprises instructions, which when executed, cause the processor to play first audio content when grouped with a portable playback device. The processor is to detect that the portable playback device is not connected to the playback network. Based on detecting that the portable playback device is not connected to the playback network, the processor is to play second audio content.

Abstract: Methods and apparatus for reducing impulsive interferences in a signal, without necessarily ascertaining a pitch frequency in the signal, detect onsets of the impulsive interferences by searching a spectrum of high-energy components for large temporal derivatives that are correlated along frequency and extend from a very low frequency up, possibly to about several kHz. The energies of the impulsive interferences are estimated, and these estimates are used to suppress the impulsive interferences. Optionally, techniques are employed to protect desired speech signals from being corrupted as a result of the suppression of the impulsive interferences.

Abstract: A dry audio signal may be received. An envelope of the dry audio signal may be compared to a threshold of the playback device to determine a dry excess. A filter may be applied to the dry audio signal to generate a wet audio signal. An envelope of the wet audio signal may be compared to the threshold to determine a wet excess. Based on the dry and wet excess, a dry gain may be determined for the dry audio signal and a wet gain may be determined for the wet audio signal. The wet audio signal and the dry audio signals may be mixed in proportion to respective gains. The mixed audio signal may avoid exceeding an operational limit associated with audio playback on the playback device.

Abstract: This is directed to controlling the output of audio based on the mode of an electronic device. In particular, this is directed to an electronic device having a sound mode, silent mode, and mute mode controlling the output of audio along different audio output routes. In the sound mode, all audio can be output, while only registered or authorized audio can be output in the silent mode. In the mute mode, no audio can be output. The sound and silent modes can be enabled using an interface of the device (e.g., a switch having two possible states), while the mute mode can be enabled using an accessory coupled to the device. To disengage the mute mode, a user can provide a corresponding instruction using the accessory, or providing an instruction on the device related to volume control. For example, a user can change the device volume using an on-device interface, or toggle a sound/silent switch to a sound mode.

Abstract: Embodiments are provided for displaying on a network device graphical display, an indication of an alarm, a media playback system playback zone associated with the alarm, and a control element for deactivating the alarm. The indication may be displayed in response to the network device receiving data indicating that the alarm is going off or is about to go off in the playback zone. A selection of the control element for deactivating the alarm may prompt a controller interface of the media playback system to be displayed on the graphical display, and the alarm may be deactivated via the controller interface. Alternatively, the selection of the control element may cause the alarm to be deactivated without displaying the controller interface. To deactivate the alarm, the network device may send to one or more playback devices in the playback zone, a transmission indicating that the alarm is to be deactivated.

Abstract: An assembly includes an analog signal processing unit. An analog signal path and an amplifier an automatic gain control unit are coupled to the signal path. Further to the output of the amplified analog signal, a gain information is provided as an analog or digital signal at a gain information output. This information can be used for further processing the signal, e.g., by automatic noise compression or to control further signal processing units.

Abstract: Systems and methods are provided to implement and facilitate cross-fading, interstitials and other effects/processing of two or more media elements in a personalized media delivery service. Effects or crossfade processing can occur on the broadcast, publisher or server-side, but can still be personalized to a specific user, in a manner that minimizes processing on the downstream side or client device. The cross-fade can be implemented after decoding the relevant chunks of each component clip, processing, recoding and rechunking, or, the cross-fade or other effect can be implemented on the relevant chunks to the effect in the compressed domain, thus obviating any loss of quality by re-encoding. A large scale personalized content delivery service can limit the processing to essentially the first and last chunks of any file, there being no need to processing the full clip.

Abstract: An apparatus and method are provided for searching content. The electronic device includes a display; an audio processing unit; and a controller configured to control the audio processing unit to output sound of first content from a foreground of the display at a preset foreground volume, detect a request to search for second content, and in response to the detected request, control the audio processing unit to simultaneously output sound of the second content from a background of the display at a preset background volume with the sound of the first content. The preset foreground volume differs from preset background volume.

Abstract: In a group level control section, a largest output level among those in channels belonging to a relevant group is selected automatically as a group level. By operating a control part of a channel level control section displayed on a basic screen, an output level of an input channel corresponding to the operated channel level control section can be controlled. Further, by operating a control part of a group level control section, output levels of the input channels belonging to the group can be collectively controlled with difference in levels among the input channels being maintained.

Abstract: Personal audio systems and methods are disclosed. A personal audio system includes a processor to generate a personal audio stream by processing an ambient audio stream in accordance with an active processing parameter set, a circular buffer memory to store a most recent snippet of the ambient audio stream, and an event detector to detect a trigger event. In response to detection of the trigger event, a controller may extract audio feature data from the most recent snippet of the ambient audio stream and transmit the audio feature data and associated metadata to a knowledgebase remote from the personal audio system.

Abstract: Systems and methods for controlling parameters of a MEMS microphone. The microphone system includes a MEMS microphone and a controller. The MEMS microphone includes a movable electrode, a stationary electrode, and a driven electrode. The movable electrode has a first side and a second side that is opposite the first side. The movable electrode is configured such that acoustic pressures acting on the first side and the second of the movable electrode cause movement of the movable electrode. The stationary electrode is positioned on the first side of the movable electrode. The driven electrode is configured to receive a control signal and alter a parameter of the MEMS microphone based on the control signal. The controller is configured to determine a voltage difference between the movable electrode and the stationary electrode. The controller is also configured to generate the control signal based on the voltage difference.

Abstract: Method for providing sound to at least one user, involves supplying audio signals from an audio signal source to a transmission unit; compressing the audio signals to generate compressed audio data; transmitting compressed audio data from the transmission unit to at least one receiver unit; decompressing the compressed audio data to generate decompressed audio signals; and stimulating the hearing of the user(s) according to decompressed audio signals supplied from the receiver unit. During certain time periods, transmission of compressed audio data is interrupted, and instead, at least one control data block is generated by the transmission unit in such a manner that audio data transmission is replaced by control data block transmission, thereby temporarily interrupting flow of received compressed audio data, each control data block includes a marker recognized by the at least one receiver unit as a control data block and a command for control of the receiver unit.

Abstract: An example playback device may store an active volume state variable in memory, where the active volume state variable corresponds to a current playback volume. The playback device may also store a volume limit state variable in memory, where the volume limit state variable corresponds to a playback volume limit of the playback device. The playback device may detect a command to begin playback of media from a given source and then select a startup playback volume to be one of the current playback volume and the playback volume limit based on the given source of the command to begin playback. The playback device may then play back media at the startup playback volume.

Abstract: An In-Car Communication (ICC) system supports the communication paths within a car by receiving the speech signals of a speaking passenger and playing it back for one or more listening passengers. Signal processing tasks are split into a microphone related part and into a loudspeaker related part. A sound processing system suitable for use in a vehicle having multiple acoustic zones includes a plurality of microphone In-Car Communication (Mic-ICC) instances coupled and a plurality of loudspeaker In-Car Communication (Ls-ICC) instances. The system further includes a dynamic audio routing matrix with a controller and coupled to the Mic-ICC instances, a mixer coupled to the plurality of Mic-ICC instances and a distributor coupled to the Ls-ICC instances.

Abstract: Technologies are generally described for modifying sound output in a personal communication device. Example devices described herein may include one or more of a communication unit, an audio processor, and/or a sound leakage detector. A communication signal may be received from the other party during personal communication by the communication unit. The communication signal may be converted by the audio processor into receiver sound for output by a receiver speaker. The sound leakage detector may provide a sound leakage indication if the sound leakage detector receives an input indicating a leakage of the receiver sound being output by the receiver speaker. Further, the audio processor may modify the receiver sound based on the sound leakage indication.

Abstract: A method of operation of a device includes receiving a first set of samples and a second set of samples. The first set of samples corresponds to a portion of a first audio frame and the second set of samples corresponds to a second audio frame. The method further includes generating a target set of samples based on the first set of samples and a first subset of the second set of samples and generating a reference set of samples based at least partially on a second subset of the second set of samples. The method also includes scaling the target set of samples to generate a scaled target set of samples and generating a third set of samples based on the scaled target set of samples and one or more samples of the second set of samples.

Abstract: Various embodiments relate to determining an energy level of one or more sound components from a sound mix. Based on at least one sound mix signal received from a mixing device and at least one component signal received from one or more sound components, at least one signal value of the sound mix signal and at least one signal value of the component signal may be computed. The component signal corresponds to each of the one or more sound components. An energy level of the one or more sound components may be determined based on the sound mix signal value and the component signal value which corresponds to each of the one or more sound components. The energy level of the one or more sound components may be output in order to determine the energy level of each component in the sound mix.

Abstract: The ambient noise headphone is a pair of headphones or a pair of earbuds that have an integral external microphone. The external microphone picks up the ambient noise external to the user and mixes this ambient noise with the primary audio source being listened to by the user. The external microphone circuitry has a volume control that allows the user to set the volume level of the ambient noise that is heard. This arrangement allows the user to monitor his external environment by controlling the amount of noise that the user can hear when listening to the pair of headphones or to the pair of earbuds. The ambient noise headphone comprises a pair of headphones, an external microphone, and a mixing circuit.

Abstract: Personal audio systems and methods are disclosed. The personal audio system includes a processor configured to generate a processed ambient audio stream by processing the ambient audio stream in accordance with a selected set of processing parameters selected from the plurality of processing parameter sets stored in the memory, a communications interface configured to receive a new set of processing parameters, selected by someone other than the user, and a controller configured to instruct the processor to begin generating the processed ambient audio stream by using the new set of processing parameters.

Abstract: An audio system for a vehicle has at least one source of audio signals. A respective directional loudspeaker array is mounted at each seat position and coupled to the at least one source. The at least one source includes a microphone that detects speech from an occupant of the first seat position. Processing circuitry receives signals from the microphone that correspond to the detect speech and drives each second respective loudspeaker array at the other seat positions to radiate acoustic energy corresponding to the detected speech.

Abstract: An example playback device may store an active volume state variable in memory, where the active volume state variable corresponds to a current playback volume. The playback device may also store a volume limit state variable in memory, where the volume limit state variable corresponds to a playback volume limit of the playback device. The playback device may detect a command to begin playback of media from a given source and then select a startup playback volume to be one of the current playback volume and the playback volume limit based on the given source of the command to begin playback. The playback device may then play back media at the startup playback volume.

Abstract: An electronic device includes a sound output circuit, a notification sound control circuit, and a stop instruction accepting circuit. The sound output circuit outputs a sound. The notification sound control circuit controls a notification sound output by the sound output circuit. The stop instruction accepting circuit accepts a stop instruction of the notification sound. The notification sound control circuit increases frequency of the notification sound in phases as time elapses when the notification sound is output by the sound output circuit, and stops the notification sound output by the sound output circuit when the instruction is accepted by the stop instruction accepting circuit.

Abstract: An entertainment system suitable for use in a vehicle and configured to control an output volume of the system includes a microphone and a processor. The microphone is configured to detect sound in a cabin of the vehicle, and output a microphone signal indicative of the sound. The processor is configured to determine a difference signal based on a difference between the microphone signal and an output signal corresponding to sound being output by the system, determine if the difference signal corresponds to human speech, and decrease the output volume of the system when the difference signal corresponds to human speech.

Abstract: A directional coding conversion method and system includes receiving input audio signals that comprise directional audio coded signals into which directional audio information is encoded according to a first loudspeaker setup and extracting the directional audio coded signals from the received input audio signals. The method and system further includes decoding, according to the first loudspeaker setup, the extracted directional audio coded signals to provide at least one absolute audio signal and corresponding absolute directional information and processing the at least one absolute audio signal and the absolute directional information to provide first output audio signals coded according to a second loudspeaker setup.

Abstract: The present invention relates to a method of protecting an inductive loudspeaker. The method comprises filtering the audio stream by applying a compensation filter to the audio stream, sending the filtered audio stream to the inductive loudspeaker, computing an estimation of a frequency response of the inductive loudspeaker and updating the compensation filter so as to attenuate a frequency corresponding to a resonant frequency in the estimated frequency response of the inductive loudspeaker.

Abstract: In accordance with systems and methods of the present disclosure, an audio device may include an electrical terminal, an audio circuit, and a transducer load detection circuit. The electrical terminal may couple a transducer device to the audio device. The audio circuit may generate an analog audio signal, wherein the analog audio signal is coupled to the electrical terminal The transducer load detection circuit may detect a load impedance of the transducer device when the transducer device is coupled to the audio device from characteristics measured at the electrical terminal.

Abstract: In accordance with systems and methods of the present disclosure, an audio device may include an electrical terminal, an audio circuit, and a transducer load detection circuit. The electrical terminal may couple a transducer device to the audio device. The audio circuit may generate an analog audio signal, wherein the analog audio signal is coupled to the electrical terminal. The transducer load detection circuit may detect a load impedance of the transducer device when the transducer device is coupled to the audio device from characteristics measured at the electrical terminal.

Abstract: Systems and techniques for processing a signal associated with a microphone are presented. The system includes a microphone component and a preamplifier. The microphone component is contained in a housing. The preamplifier includes an input buffer that receives a signal generated by the microphone component. The input buffer also generates an output signal that comprises a direct current (DC) voltage offset in comparison to the signal, where the preamplifier controls a degree of the DC voltage offset based on a control signal.

Abstract: In one example, a technique may include outputting, by a computing device associated with a user and for playback at a first volume level by an audio output device, first audio data, receiving, by the computing device, audio input data, and responsive to determining, by the computing device, that the audio input data includes speech associated with an entity different from the user, determining, by the computing device and based at least in part on the audio input data, whether to output second audio data. The method may also include, responsive to determining to output the second audio data: determining, by the computing device, a second volume level lower than the first volume level, and outputting, by the computing device and for playback at the second volume level by the audio output device, the second audio data.

Abstract: A loudness learning and balancing system for one or more audio channels. The system learns user volume preferences from manual interventions, providing effective and robust loudness learning to provide a consistent and balanced sonic experience. The system also reduces and/or eliminates incorrect attenuation or over-amplification of quiet interludes in the audio material, thereby eliminating artifacts such as noise pumping or breathing. In addition, the system reduces low level noise in the audio signals while preserving a wide dynamic range at most listening levels.

Abstract: A controller for controlling an amplification gain of an amplifier for amplifying a high dynamic range signal for an analog to digital converter comprises an input interface adapted to receive a representation of the high dynamic range signal and a signal compressor adapted to provide a low amplitude representation of the high dynamic range signal, the low amplitude representation having a lower signal amplitude than the high dynamic range signal. A comparator is adapted to compare the signal amplitude of the low amplitude representation with a predetermined threshold and an output interface is adapted to provide a control signal to the amplifier, the control signal being adapted to control the amplification gain of the amplifier such that the amplification gain is lowered when the signal amplitude of the low amplitude representation exceeds the predetermined threshold.

Abstract: An amplifier and an amplification method are provided. In accordance with the present invention, a cut-off frequency of variable high-pass filter is dynamically adjusted in real time according to a low frequency level of an input audio signal to prevent a distortion of the audio signal and damage of loudspeaker.

Abstract: An audio loudness control system includes a loudness adjuster module configured to adjust a level of an audio signal based on parameters associated with the audio signal. The parameters may include identification of a source of the audio signal. The parameters associated with the audio signal may be provided to the loudness adjuster in a message associated with the audio signal. The audio loudness control system may also include a database in communication with the loudness adjuster module. The loudness adjuster module may be configured to extract loudness related settings or parameters from the database that are associated with a source of the audio signal.

Abstract: A multiband limiter with Staggered-Y topology including a band splitter having a signal input and a plurality of bands, a first band limiter having an input coupled to a first band of the band splitter, a first summer having a first input coupled to an output of the first band limiter and a second input coupled to a second band of the band splitter, and a first summer limiter having an input coupled to an output of the first summer.

Abstract: A system includes a plurality of sources configured to provide a plurality of audio channels, and a control system including a processor, and a computer-readable storage media having computer-executable instructions embodied thereon. When executed by the processor, the computer-executable instructions cause the processor to adjust a master volume, identify a channel volume type associated with a first audio channel of the plurality of audio channels, and determine a channel volume associated with the first audio channel based at least partially on at least one of the adjustment of the master volume and the channel volume type.

Abstract: A method including the steps of assessing at least two metadata parameters associated with an audio bitstream (e.g., an encoded Dolby Digital (AC-3), Dolby Digital Plus, or Dolby E bitstream), determining individual metadata parameter quality values, each of the individual metadata parameter quality values indicative of quality (e.g., correctness) of a different one of the at least two metadata parameters, and generating data indicative of a metadata score, where the metadata score is a value determined by a combination (e.g., a linear combination or other weighted combination) of the individual metadata parameter quality values. The metadata score is indicative of overall quality (e.g., correctness) of the at least two metadata parameters. Another aspect is a system (e.g., a test device or measurement device, or another test or measurement product, or a processor) configured (e.g., programmed) to perform any embodiment of the method.

Abstract: A method of adjusting a loudness of an audio signal in real time may include receiving an electronic audio signal and dividing the audio signal into a plurality of frames. Processing of the frames may be performed in real time. The processing may include measuring initial loudness values for blocks of samples in a frame to produce a plurality of initial loudness values, and computing a weighted average of at least some of the initial loudness values. The weights may be selected based on one or more of the recency of the initial loudness values, variation of the initial loudness values, and estimated information content of the audio signal. The processing may further include selectively discarding at least some of the loudness values that reach an adaptive loudness threshold. Weights can be adjusted based on the variation of the loudness values of the audio signal.

Abstract: A class D amplifier receives and amplifies a differential analog signal which is then differentially integrated. Two pulse width modulators generate pulse signals corresponding to the differentially integrated analog signal and two power units generate output pulse signals. The outputs the power units are coupled to input terminals of integrators via a resistor feedback network. An analog output unit converts the pulse signals to an output analog signal. The differential integration circuitry implements a soft transition between mute/un-mute. In mute, the integrator output is fixed. During the soft transition, the PWM outputs change slowly from a fixed 50% duty cycle to a final value to ensure that no pop noise is present in the output as a result of mode change.

Abstract: A method and apparatus for an aircraft location system comprising an aircraft structure and a number of acoustic reflectors associated with the aircraft structure. The number of acoustic reflectors is configured to generate first sound signals in response to receiving second sound signals.

Abstract: In a speech enhancement apparatus, a generator part generates a value representing likelihood of a consonant from an input audio signal, and a calculator part generates a consonant/vowel discriminating signal for discriminating a consonant portion and a vowel portion based on the generated value, detects a first signal level of the vowel portion and a second signal level of the consonant portion based on the audio signal and the consonant/vowel discriminating signal, and outputs a level-related signal. A determining part determines a gain coefficient that exceeds one when the second signal level is smaller than the first signal level based on the level-related signal so that the gain coefficient increases as the second signal level becomes smaller than the first signal level. A multiplier part multiplies the audio signal by the gain coefficient to output an audio signal having an emphasized consonant portion.