摘要:
A sound reproduction device includes a signal processing chain configured to render an acoustic useful signal for reproduction to a listener, a simulation scenario processor configured to provide auditory scenario information for a simulated auditory scenario, the simulated auditory scenario influencing perception, by the listener, of the reproduction of the useful signal and/or defining a useful signal type, a user interface configured to detect reproduction parameter settings from a user which represent an individual preference of the listener in view of the simulated auditory scenario, a signal modifier configured to receive the reproduction parameter settings and modify reproduction of the useful signal in dependence on the reproduction parameter settings, and a storage provided for storing the reproduction parameter setting and the auditory scenario information relative to one another. Further aspects relate to a method for training user-defined and auditory scenario-dependent reproduction parameter settings for a sound reproduction device, and a corresponding computer program.
摘要:
Disclosed is a device having an audio interface configured to generate from the audio signal an outgoing audio signal for supplying to a loudspeaker component. The audio interface is configured, in generating the outgoing audio signal, to apply dynamic range compression to the audio signal. Device software is configured to receive an incoming audio signal and generate an audio signal from the incoming audio signal. The audio signal generated by the software is supplied to the audio interface for outputting by the loudspeaker component and is also used as a reference in audio signal processing. Generating the audio signal comprises the software applying initial nonlinear amplitude processing to the incoming audio signal to modify its power envelope. The modified power envelope is sufficiently smooth to be substantially unaffected by the dynamic range compression when applied by the audio interface.
摘要:
Method of dynamically adapting playback volume on electronic device starts with processor receiving first user input and first portion of audio content. First user input signals to device to increase or decrease volume of sound output. Processor determines first loudness metric corresponding to first portion of audio content when first user input is received. First loudness metric is measure of loudness of first portion of audio content being outputted. Processor then stores in memory first loudness metric in association with first user input. Memory stores history of loudness metrics in association with user inputs. Processor then determines second loudness metric that is measure of loudness of second portion of audio content that is received and determines second user input associated with second loudness metric using history. Processor generates control signal to automatically control volume of sound output by device corresponding to second user input. Other embodiments are also described.
摘要:
A directional characteristic of a microphone facility of a hearing system is more reliably controlled. The method determines a first feature value in respect of speech in a first signal of a microphone facility assigned to a first direction and a second feature value in respect of speech in a second signal of the microphone facility assigned to a second direction. A control value is obtained from the difference of the two feature values. The directional characteristic of the microphone facility is controlled with this control value.
摘要:
An exemplary system comprises a device including a memory with an audio injection application installed thereon. The application comprises an equalizer module that analyzes sound characteristics of individual digital audio samples including a discrete signal, a selector module that applies a selection heuristic to select the discrete signal from the individual digital audio samples based on the sound characteristics, and an audio module that supplies to an output an insert signal generated according to the discrete signal selected by the selection heuristic.
摘要:
A method includes identifying, at a computing device, a plurality of words in data. Each of the plurality of words corresponds to a particular word of a written language. The method includes determining a sound output level based on a location of the computing device. The method includes generating sound data based on the sound output level and the plurality of words identified in the data.
摘要:
With an acoustic device according to an embodiment, an acquiring unit acquires a frequency characteristic of external noise caused by road noise or the like and a converting unit converts a frequency characteristic of the acquired external noise to an auditory sensitivity characteristic in accordance with a frequency characteristic of auditory sensitivity. In the acoustic device, a setting unit sets a parameter that is in accordance with an auditory sensitivity characteristic in an equalizer. The equalizer corrects, in accordance with the parameter that is set by the setting unit, a frequency characteristic of an audio signal that is played back by a playback unit.
摘要:
An apparatus, method and computer program where the apparatus includes a filter configured to filter an electrical input signal and provide a filtered electrical input signal to an audio output device; an audio output device configured to convert the filtered electrical input signal to an acoustic output signal; a microphone configured to detect an acoustic signal and provide an electrical output signal corresponding to the detected acoustic signal; and a detector configured to receive the filtered electrical input signal provided to the audio output device as a first input and the electrical output signal provided by the microphone as a second input; wherein the detector is configured to detect a change in the signal power of the electrical output signal provided by the microphone relative to the filtered electrical input signal provided to the audio output device and, in response to the detection of the change in the signal power, provide a control signal to the filter to control the filter to compensate for the detected change in signal power.
摘要:
Provided is a system for adaptively enhancing an end-user's perceived quality, or quality of experience (QoE), of speech and other audio under ambient noise conditions. The system comprises the steps of determining the ambient noise characteristics on a continuous basis to capture the time varying nature of ambient noises, and adaptively determining the most optimal signal shaping to be applied to the audio/speech signal to produce the most appropriate enhancement to compensate for the ambient noise impairment. The system also comprises a signal shaping technique by using an infinite impulse response (IIR) filter that performs the signal modification with a low delay; a multi-level automatic gain control (AGC); and a controlled amplitude clipping module that assures samples are below a certain limit; and outputs the modified signal for playback through a loudspeaker or the like.
摘要:
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.